From igorolhovskiy at gmail.com Sat Sep 1 09:05:22 2018 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Sat, 1 Sep 2018 11:05:22 +0200 Subject: [Freeswitch-users] Detect Silence question Message-ID: Hi! Is there any way to make silence detection on start of the call? Like answer a call, playback a ringtone back to legA, and meanwhile try to understand, if legB is totally silent. If it is silent for, let’s say 5 sec - hangup the call. One of ways to fight spam machine calls (yes, sometimes it looks like DoS attack) Right now only thing I can imagine - record legA to file and than analyze recording for silence. Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From avi at avimarcus.net Sun Sep 2 13:47:29 2018 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 2 Sep 2018 13:47:29 +0000 Subject: [Freeswitch-users] Tagged v.1.8.1 - will there be pre-built packages, Debian 9/CentOS 7? In-Reply-To: References: <58BF4180-612E-490B-B1A3-6A58B1328F6D@jerris.com> Message-ID: <010001659a89d858-79d9f306-21f6-4f69-92c4-d9fdc938a61b-000000@email.amazonses.com> I see a deb for freeswitch 1.8 but confluence still says "debian 8 with 1.6 " and "debian 9 not yes supported ". I'm spinning up a new production system -- should I stick with debian jessie and 1.6 or is debian 9 with the latest 1.8 safe for production? Are there any breaking changes for 1.8? I can't find a human-readable changelog. Thanks! -Avi Marcus BestFone -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 3 20:13:10 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 3 Sep 2018 21:13:10 +0100 Subject: [Freeswitch-users] Add header to outbound registration Message-ID: Hello guys, Is there any way of adding a custom header to an outbound registration for a gateway? I know you can add headers on the gateway definition, but that only works for outbound calls, like this: Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 3 21:01:04 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 3 Sep 2018 22:01:04 +0100 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: Message-ID: Try &conference(YOURCONFERENCE) On Mon, Sep 3, 2018, 19:48 Do Nguyen Ha wrote: > Hi > > i read on the freeswitch website about a syntax of the originate command > > there are a lot of examples show how to use it, but there is no example to > make a new call(callA) and then bridge a call(callA) to call center > module(callcenter application) > > example to make call and bridge a call to another number > originate user/1000 &bridge(user/100) > > please help > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 3 21:02:37 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 3 Sep 2018 22:02:37 +0100 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: Message-ID: Sorry: &callcenter(default) Or your conference name. On Mon, Sep 3, 2018, 22:01 David Villasmil wrote: > Try &conference(YOURCONFERENCE) > > On Mon, Sep 3, 2018, 19:48 Do Nguyen Ha wrote: > >> Hi >> >> i read on the freeswitch website about a syntax of the originate command >> >> there are a lot of examples show how to use it, but there is no example >> to make a new call(callA) and then bridge a call(callA) to call center >> module(callcenter application) >> >> example to make call and bridge a call to another number >> originate user/1000 &bridge(user/100) >> >> please help >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at cassidywebservices.co.uk Tue Sep 4 07:57:58 2018 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 4 Sep 2018 08:57:58 +0100 Subject: [Freeswitch-users] TLS SNI In-Reply-To: References: Message-ID: No worries, thanks Brian. Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk www.cassidyweb.co.uk On Mon, 3 Sep 2018 at 18:00 Brian West wrote: > No. > > /b > > > On Thu, Aug 30, 2018 at 12:33 PM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Hi Guys, >> >> I've seen this asked before but not answered, does FreeSWITCH support SNI >> in TLS? >> >> The main reason I ask is I have a customer having issues with the Let's >> Encrypt SAN certificate I have so am looking at getting a cheap paid one >> and installing that side-by-side. >> >> Kind regards, >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS* >> Managing Director >> >> 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk >> www.cassidyweb.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Sep 4 16:32:50 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 4 Sep 2018 12:32:50 -0400 Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> Message-ID: <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely need to build these packages yourself from our support deps in stash. > On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users wrote: > > > From: kaiduan xie > Subject: mod_flite fails due to undefined symbol: cmu_lex_init > Date: August 30, 2018 at 7:10:06 PM EDT > To: FreeSWITCH Users Help > > > Hi all, > > I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, > > 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so > **/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** > > Any idea to solve this issue? > > Many thanks, > > /Kaiduan > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Sep 4 16:34:23 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 4 Sep 2018 12:34:23 -0400 Subject: [Freeswitch-users] Tagged v.1.8.1 - will there be pre-built packages, Debian 9/CentOS 7? In-Reply-To: <010001659a89d858-79d9f306-21f6-4f69-92c4-d9fdc938a61b-000000@email.amazonses.com> References: <58BF4180-612E-490B-B1A3-6A58B1328F6D@jerris.com> <010001659a89d858-79d9f306-21f6-4f69-92c4-d9fdc938a61b-000000@email.amazonses.com> Message-ID: We spent a lot of time getting new repos up last week. They are up, but still on to testing and documenting the process. 1.8.1 is currently our latest release. > On Sep 2, 2018, at 9:47 AM, Avi Marcus wrote: > > I see a deb for freeswitch 1.8 but confluence still says "debian 8 with 1.6 " and "debian 9 not yes supported ". > > I'm spinning up a new production system -- should I stick with debian jessie and 1.6 or is debian 9 with the latest 1.8 safe for production? > > Are there any breaking changes for 1.8? I can't find a human-readable changelog. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From matthew at brightfire.net Tue Sep 4 16:45:40 2018 From: matthew at brightfire.net (Matthew Grooms) Date: Tue, 4 Sep 2018 11:45:40 -0500 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: Message-ID: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> To originate and connect directly to mod_callcenter, you should be able to just call the application with the queue name ... originate orgination-string &callcenter(queue-name) for example: orginate sofia/gateway/mygateway/+15555555555 &callcenter(1000 at default) It sounds like your trying to do outbound call center dialing. I have an extension configured for each call center queue. My external dial manager calls into freeswitch to launch a lua script with a bunch of arguments. That script does an originate, checks if a person or a machine answered ( AMD module + a bunch of other checks ) and then transfers people to the extension associated with the call center queue. It's not difficult with a little trial and error. The hard part is building the dial manger logic :/ Hope this helps, -Matthew On 8/31/2018 12:32 PM, Do Nguyen Ha wrote: > Hi > > i read on the freeswitch website about a syntax of the originate command > > there are a lot of examples show how to use it, but there is no > example to make a new call(callA) and then bridge a call(callA) to > call center module(callcenter application) > > example to make call and bridge a call to another number > originate user/1000 &bridge(user/100) > > please help > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Sep 4 18:37:12 2018 From: brian at freeswitch.com (Brian West) Date: Tue, 4 Sep 2018 13:37:12 -0500 Subject: [Freeswitch-users] TLS SNI In-Reply-To: References: Message-ID: You can email sales at freeswitch.com and engage our commercial solutions team and see if we can do this. Thanks, /b On Tue, Sep 4, 2018 at 2:57 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > No worries, thanks Brian. > > Kind regards, > > -- > *Andrew Cassidy BSc (Hons) MBCS* > Managing Director > > 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk ww > w.cassidyweb.co.uk > > On Mon, 3 Sep 2018 at 18:00 Brian West wrote: > >> No. >> >> /b >> >> >> On Thu, Aug 30, 2018 at 12:33 PM, Andrew Cassidy < >> andrew at cassidywebservices.co.uk> wrote: >> >>> Hi Guys, >>> >>> I've seen this asked before but not answered, does FreeSWITCH support >>> SNI in TLS? >>> >>> The main reason I ask is I have a customer having issues with the Let's >>> Encrypt SAN certificate I have so am looking at getting a cheap paid one >>> and installing that side-by-side. >>> >>> Kind regards, >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS* >>> Managing Director >>> >>> 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk ww >>> w.cassidyweb.co.uk >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Tue Sep 4 19:45:41 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 4 Sep 2018 14:45:41 -0500 Subject: [Freeswitch-users] Phone spontaneously going into Hold when a call is established? Message-ID: I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside number was going immediately to 'On Hold' when the person answered. Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. -- Nathan  ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Wed Sep 5 01:05:22 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Wed, 5 Sep 2018 08:05:22 +0700 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: Message-ID: thank you it works does callcenter support any args as variables?? Thank you On Tue, Sep 4, 2018 at 11:15 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Sorry: > &callcenter(default) > Or your conference name. > > On Mon, Sep 3, 2018, 22:01 David Villasmil > wrote: > >> Try &conference(YOURCONFERENCE) >> >> On Mon, Sep 3, 2018, 19:48 Do Nguyen Ha wrote: >> >>> Hi >>> >>> i read on the freeswitch website about a syntax of the originate command >>> >>> there are a lot of examples show how to use it, but there is no example >>> to make a new call(callA) and then bridge a call(callA) to call center >>> module(callcenter application) >>> >>> example to make call and bridge a call to another number >>> originate user/1000 &bridge(user/100) >>> >>> please help >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Wed Sep 5 01:24:12 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Wed, 5 Sep 2018 08:24:12 +0700 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> References: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> Message-ID: thank you for sharing your experience my goal is simple auto calling from esl to a list of numbers(callers) when callers answer phone and then i will bridge to callcenter module to check the callers answer or not of originate command, i use the event-name "BACKGROUND_JOB". the message body to check is "_body": "-ERR USER_BUSY\n"/"-ERR NO_ANSWER\n"/"-ERR DESTINATION_OUT_OF_ORDER\n"... to check the callers is answered by the agents of callcenter i use the event-name "CUSTOM" and event-name "bridge-agent-start" that is my dial manger logic is there any suggestion thank you On Wed, Sep 5, 2018 at 2:19 AM Matthew Grooms wrote: > To originate and connect directly to mod_callcenter, you should be able to > just call the application with the queue name ... > > originate orgination-string &callcenter(queue-name) > > for example: > orginate sofia/gateway/mygateway/+15555555555 &callcenter(1000 at default) > > It sounds like your trying to do outbound call center dialing. I have an > extension configured for each call center queue. My external dial manager > calls into freeswitch to launch a lua script with a bunch of arguments. > That script does an originate, checks if a person or a machine answered ( > AMD module + a bunch of other checks ) and then transfers people to the > extension associated with the call center queue. It's not difficult with a > little trial and error. The hard part is building the dial manger logic :/ > > Hope this helps, > > -Matthew > On 8/31/2018 12:32 PM, Do Nguyen Ha wrote: > > Hi > > i read on the freeswitch website about a syntax of the originate command > > there are a lot of examples show how to use it, but there is no example to > make a new call(callA) and then bridge a call(callA) to call center > module(callcenter application) > > example to make call and bridge a call to another number > originate user/1000 &bridge(user/100) > > please help > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 5 11:33:09 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 5 Sep 2018 12:33:09 +0100 Subject: [Freeswitch-users] fs_cli double free or corruption. Message-ID: Hello all, just FYI: working on fs_cli, after doing an bgapi originate i got *** Error in `fs_cli': double free or corruption (fasttop): 0x0000000001283070 *** Don't have a dump, sorry. freeswitch at argo-fs04.singlecomm.com> status UP 0 years, 0 days, 9 hours, 30 minutes, 1 second, 19 milliseconds, 690 microseconds FreeSWITCH (Version 1.6.19 -36-7a77e0b 64bit) is ready 373 session(s) since startup 0 session(s) - peak 21, last 5min 2 0 session(s) per Sec out of max 30, peak 5, last 5min 1 1000 session(s) max min idle cpu 0.00/98.40 Current Stack Size/Max 240K/8192K Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rw at panorgan.ch Wed Sep 5 11:56:21 2018 From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=) Date: Wed, 5 Sep 2018 13:56:21 +0200 Subject: [Freeswitch-users] Show ring-group name on phone Message-ID: Hi What is the best/right way to show the name of the called ring-group on the ringing phone? At the moment I have configured 2 ring groups (let's call them "main number" and "hotline") which bridge incoming calls like this: The problem is now, that on the display of the ringing phones I can only see that there's a call from to the number of the individual phone. What I would like to see is: call from to (or ) (We use Yealink phones (mostly T46s) if that helps) Thanks. From t90fpe at outlook.com Wed Sep 5 12:05:05 2018 From: t90fpe at outlook.com (Fred Pettersson) Date: Wed, 5 Sep 2018 12:05:05 +0000 Subject: [Freeswitch-users] Capture the audio stream of a call once answered? Message-ID: Hi, is it possible to get hold of/capture the "voice stream" (the RTP stream I guess) once you answered the call? I would like to send it to an external server with our own service for processing the stream but we would like to keep control of the answered call in FreeSWITCH. /Fred -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Sep 5 13:55:40 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 5 Sep 2018 14:55:40 +0100 Subject: [Freeswitch-users] SIP calls in from provider Message-ID: I have a provider wanting to route calls in via direct SIP address. I’m not sure exactly what’s needed as used to setting up a gateway with username and password. Can anyone summarise (or point me to) the steps required to make this work? Am thinking: - ACL for provider’s IPs - Something needed in ‘public’ dial plan to catch whateveraddress at mysipdomain.com - what exactly? - Anything else required? -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Wed Sep 5 14:13:49 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Wed, 5 Sep 2018 14:13:49 +0000 (UTC) Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> Message-ID: <117989109.1724659.1536156829596@mail.yahoo.com> Thanks Michael. I downloaded the flite 2.0 from http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz and build it as below ./configure --enable-sharedmakemake install build and run freeswitch-1.8, the same error occurred again. However mod_flite was successfully loaded if I ran freeswitch as below (used LD_PRELOAD to load libflite_cmulex) kxie at kxie-Inspiron-3656:~/freeswitch-install$ pwd/home/kxie/freeswitch-installkxie at kxie-Inspiron-3656:~/freeswitch-install$ LD_PRELOAD=/usr/local/lib/libflite_cmulex.so ./bin/freeswitch /Kaiduan On Tuesday, September 4, 2018, 12:32:54 PM EDT, Michael Jerris wrote: Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely need to build these packages yourself from our support deps in stash. On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users wrote: From: kaiduan xie Subject: mod_flite fails due to undefined symbol: cmu_lex_init Date: August 30, 2018 at 7:10:06 PM EDT To: FreeSWITCH Users Help Hi all, I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so**/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** Any idea to solve this issue? Many thanks, /Kaiduan _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From matthew at brightfire.net Wed Sep 5 15:50:32 2018 From: matthew at brightfire.net (Matthew Grooms) Date: Wed, 5 Sep 2018 10:50:32 -0500 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> Message-ID: On 9/4/2018 8:24 PM, Do Nguyen Ha wrote: > thank you for sharing your experience > > my goal is simple auto calling from esl to a list of numbers(callers) > when callers answer phone and then i will bridge to callcenter module > > to check the callers answer or not of originate command, i use the > event-name "BACKGROUND_JOB". the message body to check is "_body": > "-ERR USER_BUSY\n"/"-ERR NO_ANSWER\n"/"-ERR DESTINATION_OUT_OF_ORDER\n"... > I believe those are based on the SIP response codes. You should check out the commercial AMD module which will help you detect if you've reached a human or a machine ... https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd > to check the callers is answered by the agents of callcenter i use the > event-name "CUSTOM" and event-name  "bridge-agent-start" > > that is my dial manger logic > > is there any suggestion > Sounds like your on the right path. But in my experience, dialer logic has less to do with overcoming technical challenges and more to do with adhering to the legal requirements associated with outbound dialing. -Matthew From kaiduanx at yahoo.ca Wed Sep 5 21:52:40 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Wed, 5 Sep 2018 21:52:40 +0000 (UTC) Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> Message-ID: <1425516003.163153.1536184360277@mail.yahoo.com> With the command as below, LD_PRELOAD="/usr/local/lib/libflite_cmu_us_kal.so /usr/local/lib/libflite_usenglish.so /usr/local/lib/libflite_cmulex.so /usr/local/lib/libflite.so" ./bin/freeswitch I finally got mod_flite working on Ubuntu 16.04 with Freeswitch 1.8.0 source code So the question is why Freeswitch does not load above libraries by itself? /Kaiduan On Wednesday, September 5, 2018, 11:54:07 AM EDT, kaiduan xie via FreeSWITCH-users wrote: Thanks Michael. I downloaded the flite 2.0 from http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz and build it as below ./configure --enable-sharedmakemake install build and run freeswitch-1.8, the same error occurred again. However mod_flite was successfully loaded if I ran freeswitch as below (used LD_PRELOAD to load libflite_cmulex) kxie at kxie-Inspiron-3656:~/freeswitch-install$ pwd/home/kxie/freeswitch-installkxie at kxie-Inspiron-3656:~/freeswitch-install$ LD_PRELOAD=/usr/local/lib/libflite_cmulex.so ./bin/freeswitch /Kaiduan On Tuesday, September 4, 2018, 12:32:54 PM EDT, Michael Jerris wrote: Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely need to build these packages yourself from our support deps in stash. On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users wrote: From: kaiduan xie Subject: mod_flite fails due to undefined symbol: cmu_lex_init Date: August 30, 2018 at 7:10:06 PM EDT To: FreeSWITCH Users Help Hi all, I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so**/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** Any idea to solve this issue? Many thanks, /Kaiduan _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Sep 6 09:04:50 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Sep 2018 05:04:50 -0400 Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: <1425516003.163153.1536184360277@mail.yahoo.com> References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> <1425516003.163153.1536184360277@mail.yahoo.com> Message-ID: <61573118-9C8A-42D0-B680-1EBCCCFFEECE@jerris.com> I don’t see this issue on debian. ld.so.conf might fix it, or building to /usr instead of /usr/local. Check out our debian directory for how we build the packages for debian as a hint. On debian 9 we just started using system packages as they now have working flite 2.0 in the distro, see master for the patches necessary for that to work without pkg-config (which unfortunately the debian package does not have) Our deb directory is here: https://freeswitch.org/stash/projects/SD/repos/libflite/browse/debian You might want to use the src from this git checkout as I’m not sure the tarball has our debian packaging in it or not. Mike > On Sep 5, 2018, at 5:52 PM, kaiduan xie wrote: > > With the command as below, > > LD_PRELOAD="/usr/local/lib/libflite_cmu_us_kal.so /usr/local/lib/libflite_usenglish.so /usr/local/lib/libflite_cmulex.so /usr/local/lib/libflite.so" ./bin/freeswitch > > I finally got mod_flite working on Ubuntu 16.04 with Freeswitch 1.8.0 source code > > So the question is why Freeswitch does not load above libraries by itself? > > /Kaiduan > On Wednesday, September 5, 2018, 11:54:07 AM EDT, kaiduan xie via FreeSWITCH-users wrote: > > > Thanks Michael. > > I downloaded the flite 2.0 from http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz and build it as below > > ./configure --enable-shared > make > make install > > build and run freeswitch-1.8, the same error occurred again. > > However mod_flite was successfully loaded if I ran freeswitch as below (used LD_PRELOAD to load libflite_cmulex) > > kxie at kxie-Inspiron-3656:~/freeswitch-install$ pwd > /home/kxie/freeswitch-install > kxie at kxie-Inspiron-3656:~/freeswitch-install$ LD_PRELOAD=/usr/local/lib/libflite_cmulex.so ./bin/freeswitch > > /Kaiduan > > On Tuesday, September 4, 2018, 12:32:54 PM EDT, Michael Jerris wrote: > > > Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely need to build these packages yourself from our support deps in stash. > >> On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users > wrote: >> >> >> From: kaiduan xie > >> Subject: mod_flite fails due to undefined symbol: cmu_lex_init >> Date: August 30, 2018 at 7:10:06 PM EDT >> To: FreeSWITCH Users Help > >> >> >> Hi all, >> >> I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, >> >> 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so >> **/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** >> >> Any idea to solve this issue? >> >> Many thanks, >> >> /Kaiduan >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Thu Sep 6 13:26:21 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 6 Sep 2018 08:26:21 -0500 Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: <61573118-9C8A-42D0-B680-1EBCCCFFEECE@jerris.com> References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> <1425516003.163153.1536184360277@mail.yahoo.com> <61573118-9C8A-42D0-B680-1EBCCCFFEECE@jerris.com> Message-ID: <276dc431-3cb3-513f-a739-55bf9057e9d4@mst.edu> kaiduan - when you _built_ freeswitch, did you make sure it built using the extra libs you installed manually? I do custom builds of all of the dependent libs on my box (ubuntu 16), and install into /local/fslibs/install, and then during build I use:     export PKG_CONFIG_PATH=/local/fslibs/install/lib/pkgconfig:/usr/share/pkgconfig     cd /local/freeswitch/src     ./configure -C --disable-fhs --prefix=/local/freeswitch/server At that point I don't have to set any env vars to get FS to launch. -- Nathan ------------------------------------------------------------------------------------------------------------------------ *From:* Michael Jerris *Sent:* Thu, Sep 6, 2018 4:04 AM CDT *To:* FreeSWITCH Users Help *Subject:* [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init > I don’t see this issue on debian.  ld.so.conf might fix it, or building to /usr instead of /usr/local.  Check out our > debian directory for how we build the packages for debian as a hint.  On debian 9 we just started using system > packages as they now have working flite 2.0 in the distro, see master for the patches necessary for that to work > without pkg-config (which unfortunately the debian package does not have) > > Our deb directory is here: https://freeswitch.org/stash/projects/SD/repos/libflite/browse/debian > > You might want to use the src from this git checkout as I’m not sure the tarball has our debian packaging in it or not. > > Mike > > > >> On Sep 5, 2018, at 5:52 PM, kaiduan xie > wrote: >> >> With the command as below, >> >> LD_PRELOAD="/usr/local/lib/libflite_cmu_us_kal.so /usr/local/lib/libflite_usenglish.so >> /usr/local/lib/libflite_cmulex.so /usr/local/lib/libflite.so" ./bin/freeswitch >> >> I finally got mod_flite working on Ubuntu 16.04 with Freeswitch 1.8.0 source code >> >> So the question is why Freeswitch does not load above libraries by itself? >> >> /Kaiduan >> On Wednesday, September 5, 2018, 11:54:07 AM EDT, kaiduan xie via FreeSWITCH-users >> > wrote: >> >> >> Thanks Michael. >> >> I downloaded the flite 2.0 from http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz and build it as below >> >> ./configure --enable-shared >> make >> make install >> >> build and run freeswitch-1.8, the same error occurred again. >> >> However mod_flite was successfully loaded if I ran freeswitch as below (used LD_PRELOAD to load libflite_cmulex) >> >> kxie at kxie-Inspiron-3656:~/freeswitch-install$ pwd >> /home/kxie/freeswitch-install >> kxie at kxie-Inspiron-3656:~/freeswitch-install$ LD_PRELOAD=/usr/local/lib/libflite_cmulex.so ./bin/freeswitch >> >> /Kaiduan >> >> On Tuesday, September 4, 2018, 12:32:54 PM EDT, Michael Jerris > wrote: >> >> >> Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely >> need to build these packages yourself from our support deps in stash. >> >>> On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users >> > wrote: >>> >>> >>> *From: *kaiduan xie > >>> *Subject: **mod_flite fails due to undefined symbol: cmu_lex_init* >>> *Date: *August 30, 2018 at 7:10:06 PM EDT >>> *To: *FreeSWITCH Users Help > >>> >>> >>> Hi all, >>> >>> I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, >>> >>> 2018-08-30 19:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module >>> /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so >>> **/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** >>> >>> Any idea to solve this issue? >>> >>> Many thanks, >>> >>> /Kaiduan >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefano.bertuola at gmail.com Thu Sep 6 11:35:14 2018 From: stefano.bertuola at gmail.com (Stefano Bertuola) Date: Thu, 6 Sep 2018 13:35:14 +0200 Subject: [Freeswitch-users] =?utf-8?q?=28no_subject=29?= Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Thu Sep 6 18:34:14 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 6 Sep 2018 18:34:14 +0000 (UTC) Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init In-Reply-To: <276dc431-3cb3-513f-a739-55bf9057e9d4@mst.edu> References: <1075013395.1371073.1535670606654.ref@mail.yahoo.com> <3BE057B5-8C73-43A8-ADBF-1299ECEB5E74@jerris.com> <1425516003.163153.1536184360277@mail.yahoo.com> <61573118-9C8A-42D0-B680-1EBCCCFFEECE@jerris.com> <276dc431-3cb3-513f-a739-55bf9057e9d4@mst.edu> Message-ID: <399713140.763235.1536258854758@mail.yahoo.com> Nathan, After build and install flite to /usr/local, I build freeswitch as below, 1. ./devel-bootstrap.sh --prefix=/home/kxie/freeswitch-dev-install2. change modules.conf to enable flite3. make4. make install /Kaiduan On Thursday, September 6, 2018, 2:02:20 PM EDT, Nathan Neulinger wrote: kaiduan - when you _built_ freeswitch, did you make sure it built using the extra libs you installed manually? I do custom builds of all of the dependent libs on my box (ubuntu 16), and install into /local/fslibs/install, and then during build I use:     export PKG_CONFIG_PATH=/local/fslibs/install/lib/pkgconfig:/usr/share/pkgconfig     cd /local/freeswitch/src     ./configure -C --disable-fhs --prefix=/local/freeswitch/server At that point I don't have to set any env vars to get FS to launch. -- Nathan From: Michael Jerris Sent: Thu, Sep 6, 2018 4:04 AM CDT To: FreeSWITCH Users Help Subject: [Freeswitch-users] mod_flite fails due to undefined symbol: cmu_lex_init I don’t see this issue on debian.  ld.so.conf might fix it, or building to /usr instead of /usr/local.  Check out our debian directory for how we build the packages for debian as a hint.  On debian 9 we just started using system packages as they now have working flite 2.0 in the distro, see master for the patches necessary for that to work without pkg-config (which unfortunately the debian package does not have) Our deb directory is here:  https://freeswitch.org/stash/projects/SD/repos/libflite/browse/debian You might want to use the src from this git checkout as I’m not sure the tarball has our debian packaging in it or not. Mike On Sep 5, 2018, at 5:52 PM, kaiduan xie wrote: With the command as below, LD_PRELOAD="/usr/local/lib/libflite_cmu_us_kal.so /usr/local/lib/libflite_usenglish.so /usr/local/lib/libflite_cmulex.so /usr/local/lib/libflite.so" ./bin/freeswitch I finally got mod_flite working on Ubuntu 16.04 with Freeswitch 1.8.0 source code So the question is why Freeswitch does not load above libraries by itself? /Kaiduan On Wednesday, September 5, 2018, 11:54:07 AM EDT, kaiduan xie via FreeSWITCH-users wrote: Thanks Michael. I downloaded the flite 2.0 from http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz and build it as below ./configure --enable-shared make make install build and run freeswitch-1.8, the same error occurred again. However mod_flite was successfully loaded if I ran freeswitch as below (used LD_PRELOAD to load libflite_cmulex) kxie at kxie-Inspiron-3656:~/freeswitch-install$ pwd /home/kxie/freeswitch-install kxie at kxie-Inspiron-3656:~/freeswitch-install$LD_PRELOAD=/usr/local/lib/libflite_cmulex.so ./bin/freeswitch /Kaiduan On Tuesday, September 4, 2018, 12:32:54 PM EDT, Michael Jerris wrote: Our build requires flite 2.0 …. Our latest Debian unstable and 1.8 repos have these packages, for ubuntu you likely need to build these packages yourself from our support deps in stash. On Sep 3, 2018, at 12:26 PM, kaiduan xie via FreeSWITCH-users wrote: From: kaiduan xie Subject: mod_flite fails due to undefined symbol: cmu_lex_init Date: August 30, 2018 at 7:10:06 PM EDT To: FreeSWITCH Users Help Hi all, I build and run FreeSwitch 1.6.20 from source on Ubuntu 16.0.4 xenial, the mod_filte failed while loading, 2018-08-3019:03:30.786468 [CRIT] switch_loadable_module.c:1522 Error Loading module /home/kxie/freeswitch-dev-install/lib/freeswitch/mod/mod_flite.so **/usr/local/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** Any idea to solve this issue? Many thanks, /Kaiduan _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mario_fs at mgtech.com Thu Sep 6 19:22:40 2018 From: mario_fs at mgtech.com (Mario) Date: Thu, 6 Sep 2018 12:22:40 -0700 Subject: [Freeswitch-users] 1.8.1 zip missing ./configure Message-ID: <82CC17F4-B5CD-4FE4-BC71-FD8D671E546F@mgtech.com> Just saw 1.8.1 show up in files.freeswitch.org . I tried to build from .zip but ./configure is missing, not sure if this is a bug or design change. INSTALL file still says use ./configure. Mario G -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Fri Sep 7 03:41:44 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Fri, 7 Sep 2018 10:41:44 +0700 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> Message-ID: Thank you for your comments On Thu, Sep 6, 2018 at 12:01 AM Matthew Grooms wrote: > On 9/4/2018 8:24 PM, Do Nguyen Ha wrote: > > thank you for sharing your experience > > > > my goal is simple auto calling from esl to a list of numbers(callers) > > when callers answer phone and then i will bridge to callcenter module > > > > to check the callers answer or not of originate command, i use the > > event-name "BACKGROUND_JOB". the message body to check is "_body": > > "-ERR USER_BUSY\n"/"-ERR NO_ANSWER\n"/"-ERR > DESTINATION_OUT_OF_ORDER\n"... > > > > I believe those are based on the SIP response codes. You should check > out the commercial AMD module which will help you detect if you've > reached a human or a machine ... > > https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd > > > to check the callers is answered by the agents of callcenter i use the > > event-name "CUSTOM" and event-name "bridge-agent-start" > > > > that is my dial manger logic > > > > is there any suggestion > > > > Sounds like your on the right path. But in my experience, dialer logic > has less to do with overcoming technical challenges and more to do with > adhering to the legal requirements associated with outbound dialing. > > -Matthew > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Fri Sep 7 03:48:09 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 6 Sep 2018 22:48:09 -0500 Subject: [Freeswitch-users] Phone spontaneously going into Hold when a call is established? In-Reply-To: References: Message-ID: Finally managed to get a reproducible failure on this, but don't understand SIP enough to know where it's going wrong. I have two local numbers that I can call via my local telco over a SIP trunk (MetaSwitch) - one reliably demonstrates the behavior, the other reliably works normal: The call that is working normally (leg w/ provider): INVITE w/ neither sendonly nor sendrecv 100 Trying 183 Progress w/ a=sendrecv 200 OK w/ a=sendrecv The call that is going into on-hold immediately (leg w/ provider): INVITE w/ neither sendonly nor sendrecv 100 Trying 180 Ringing 200 OK w/ a=sendonly INVITE w/ a=sendrecv 100 Trying 200 OK w/ neither sendonly nor sendrecv Call that goes into on hold immediately (leg w/ device): INVITE w/ neither, but has sdp 407 INVITE w/ neither, but has sdp 100 183 w/ neither 200 w/ neither UPDATE w/ sendonly (which is after the second 200 OK above) 200 OK w/ recvonly I'm going to try and do some more digging on this tomorrow, but if someone has any insights here on what the proper/expected behavior is, I sure would love to hear it. It seems like this may be somewhat related to FS-9765 - however there is no invite w/o an SDP. Also similar sounding to FS-11234. I'm happy to take this forward to provider (I'm inquiring with them as well) but would be helpful to have some more details on which piece is behaving incorrectly. -- Nathan -------- Original Message -------- From: Nathan Neulinger Sent: Tue, Sep 4, 2018 2:45 PM CDT To: FreeSWITCH Users Help Subject: Phone spontaneously going into Hold when a call is established? I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside number was going immediately to 'On Hold' when the person answered. Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. -- Nathan  ------------------------------------------------------------ Nathan Neulingernneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From alex at freeswitch.com Fri Sep 7 06:11:23 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 7 Sep 2018 15:11:23 +0900 Subject: [Freeswitch-users] SIP calls in from provider In-Reply-To: References: Message-ID: You need only two things: 1. Adjust your public context to catch calls to your number and transfer them to default (example is in vanilla config already) 2. Ask your provider to call to 5080 port That's should be enough. Regards, Alex On Wed, Sep 5, 2018 at 11:24 PM Rick Jarvis wrote: > I have a provider wanting to route calls in via direct SIP address. I’m > not sure exactly what’s needed as used to setting up a gateway with > username and password. > > Can anyone summarise (or point me to) the steps required to make this work? > > Am thinking: > > - ACL for provider’s IPs > - Something needed in ‘public’ dial plan to catch > whateveraddress at mysipdomain.com - what exactly? > - Anything else required? > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Sep 7 06:15:46 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 7 Sep 2018 15:15:46 +0900 Subject: [Freeswitch-users] Show ring-group name on phone In-Reply-To: References: Message-ID: Hi, Try something like this: Regards, Alex On Wed, Sep 5, 2018 at 11:33 PM René Weiss wrote: > Hi > > What is the best/right way to show the name of the called ring-group > on the ringing phone? > > At the moment I have configured 2 ring groups (let's call them "main > number" and "hotline") which bridge incoming calls like this: > > data="user/101@$${domain_name}, > user/102@$${domain_name}, > user/103@$${domain_name} > "/> > > The problem is now, that on the display of the ringing phones > I can only see that there's a call from to the > number of the individual phone. > > What I would like to see is: call from to > (or ) > > (We use Yealink phones (mostly T46s) if that helps) > > Thanks. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From leevidor89 at googlemail.com Fri Sep 7 14:50:12 2018 From: leevidor89 at googlemail.com (Lee Vidor) Date: Fri, 7 Sep 2018 15:50:12 +0100 Subject: [Freeswitch-users] mod_tts_commandline hangs in call flow In-Reply-To: References: Message-ID: It seems enabling the following command resolves my problem: Can anyone explain in detail what this variable does? Thanks! On Mon, 3 Sep 2018 at 20:30, Lee Vidor via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Lee Vidor > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Fri, 31 Aug 2018 13:50:27 +0100 > Subject: mod_tts_commandline hangs in call flow > Hi > > I'm currently using mod_tts_commandline with a python script to AWS text > to speech API. The flow works fine during a single call. > > Event socket API -> Freeswitch dialplan -> mod_tts_commandline -> Python > script -> AWS API -> Voice file -> Freeswitch say > > However when there a number of concurrent calls (even just 10 to 15) the > request to execute the python script hangs until all the calls are at the > same requesting state. As if mod_tts_commandline is waiting for something > to happen. > > Dial plan: > > > > > > > data="tts_commandline|${lang}|${text}" /> > > > > > tts_commandline.conf: > > > > > Execution of python script in freeswitch logs: > > 2018-08-30 11:19:57.641892 [ERR] switch_ivr_play_say.c:2385 > switch_ivr_detect_silence: SILENCE DETECTED > 2018-08-30 11:19:57.641892 [DEBUG] switch_core_codec.c:248 > sofia/external/33123456789 Restore previous codec PCMU:0. > EXECUTE sofia/external/33123456789 speak(tts_commandline|en-US|Hi, This is > a test message) > 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3025 OPEN TTS > tts_commandline > 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3035 Raw Codec > Activated > 2018-08-30 11:19:57.641892 [DEBUG] mod_tts_commandline.c:160 Executing: > /usr/bin/python /usr/local/freeswitch/scripts/synthesize_text.py > --language='en-US' > --filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav' --text='Hi, > This is a test message' > > "Simultaneous calls requesting mod_tts_commandline" > > Log from python script: (notice it's called with a 15 second delay) ---> *What > is causing this delay?* > > 2018-08-30 11:20:12,488 DEBUG Called script > 2018-08-30 11:20:12,490 DEBUG Arguments > Namespace(filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav', > language='en-US', ssml=None, text='Hi, This is a test message') > 2018-08-30 11:20:12,490 DEBUG synthesize_text requested > > The python script executes fine and returns to freeswitch: > > 2018-08-30 11:20:14.541919 [DEBUG] switch_core_file.c:342 File > /tmp/c61d2034-c1b1-4f42-932f-350484553b5a.tmp.wav sample rate 24000 doesn't > match requested rate 8000 > 2018-08-30 11:20:14.541919 [DEBUG] switch_ivr_play_say.c:2729 Speaking > text: Hi, This is a test message > > Any help or further debugging steps would be appreciated. > > Thanks > > Lee > > > > ---------- Forwarded message ---------- > From: Lee Vidor via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 03 Sep 2018 12:30:09 -0700 (PDT) > Subject: [Freeswitch-users] mod_tts_commandline hangs in call flow > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Fri Sep 7 14:56:08 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 7 Sep 2018 09:56:08 -0500 Subject: [Freeswitch-users] Phone spontaneously going into Hold when a call is established? In-Reply-To: References: Message-ID: Went through a git bisect on this, and tracked the behavior change to this commit; commit 40bfe0fff566476ff95242581f391f5b5b1c7e32 Author: Anthony Minessale Date:   Mon Jan 16 15:27:36 2017 -0600     FS-9154: [freeswitch-core] Add & remove video on re-invites #resolve Note that these calls do NOT have any video. I've added this info to FS-9983 jira issue which looks highly related, but will go ahead and file a new jira issue as well. -- Nathan * * *From:* Nathan Neulinger *Sent:* Thu, Sep 6, 2018 10:48 PM CDT *To:* FreeSWITCH Users Help *Subject:* Phone spontaneously going into Hold when a call is established? > Finally managed to get a reproducible failure on this, but don't understand SIP enough to know where it's going wrong. > I have two local numbers that I can call via my local telco over a SIP trunk (MetaSwitch) - one reliably demonstrates > the behavior, the other reliably works normal: > > The call that is working normally (leg w/ provider): >     INVITE w/ neither sendonly nor sendrecv >     100 Trying >     183 Progress w/ a=sendrecv >     200 OK w/ a=sendrecv > > The call that is going into on-hold immediately (leg w/ provider): >     INVITE w/ neither sendonly nor sendrecv >     100 Trying >     180 Ringing >     200 OK  w/ a=sendonly >     INVITE w/ a=sendrecv >     100 Trying >     200 OK w/ neither sendonly nor sendrecv > > Call that goes into on hold immediately (leg w/ device): >     INVITE w/ neither, but has sdp >     407 >     INVITE w/ neither, but has sdp >     100 >     183 w/ neither >     200 w/ neither >     UPDATE w/ sendonly (which is after the second 200 OK above) >     200 OK w/ recvonly > > > > I'm going to try and do some more digging on this tomorrow, but if someone has any insights here on what the > proper/expected behavior is, I sure would love to hear it. > > It seems like this may be somewhat related to FS-9765 - however there is no invite w/o an SDP. Also similar sounding > to FS-11234. I'm happy to take this forward to provider (I'm inquiring with them as well) but would be helpful to have > some more details on which piece is behaving incorrectly. > > -- Nathan > > -------- Original Message -------- > From: Nathan Neulinger > Sent: Tue, Sep 4, 2018 2:45 PM CDT > To: FreeSWITCH Users Help > Subject: Phone spontaneously going into Hold when a call is established? > > I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside > number was going immediately to 'On Hold' when the person answered. > > Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a > polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> > 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. > > -- Nathan >   ------------------------------------------------------------ > Nathan Neulingernneul at mst.edu > Missouri S&T Information Technology    (573) 612-1412 > System Administrator - Architect > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 7 18:02:36 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 7 Sep 2018 14:02:36 -0400 Subject: [Freeswitch-users] 1.8.1 zip missing ./configure In-Reply-To: <82CC17F4-B5CD-4FE4-BC71-FD8D671E546F@mgtech.com> References: <82CC17F4-B5CD-4FE4-BC71-FD8D671E546F@mgtech.com> Message-ID: <92FD0269-5BD1-4A74-9299-2034235F5066@jerris.com> Its a bug. You can just run bootstrap to get one. We will correct this issue in release files. > On Sep 6, 2018, at 3:22 PM, Mario wrote: > > Just saw 1.8.1 show up in files.freeswitch.org . I tried to build from .zip but ./configure is missing, not sure if this is a bug or design change. INSTALL file still says use ./configure. > Mario G -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 7 18:09:04 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 7 Sep 2018 14:09:04 -0400 Subject: [Freeswitch-users] Phone spontaneously going into Hold when a call is established? In-Reply-To: References: Message-ID: <72FBA931-44C1-4796-AFD0-A4956958FF1B@jerris.com> Some notes here… if there is none of the inactive/sendonly/recvonly/sendrecv its the same as if it was sendrecv In the going on to on-hold immediately (leg w/ provider) case… the call is being answered sendonly, essentially they are answering the call on hold, then sending us a re-invite to take us off hold. Its correct that the call should start on hold, but it should come back off hold, can you confirm that it is? In the going on to on-hold immediately (leg w/ device) case… the call is being answered off hold, and then we are getting an update with sendonly (putting us on hold)…. I would expect it to be on hold after that. > On Sep 6, 2018, at 11:48 PM, Nathan Neulinger wrote: > > Finally managed to get a reproducible failure on this, but don't understand SIP enough to know where it's going wrong. I have two local numbers that I can call via my local telco over a SIP trunk (MetaSwitch) - one reliably demonstrates the behavior, the other reliably works normal: > > The call that is working normally (leg w/ provider): > INVITE w/ neither sendonly nor sendrecv > 100 Trying > 183 Progress w/ a=sendrecv > 200 OK w/ a=sendrecv > > The call that is going into on-hold immediately (leg w/ provider): > INVITE w/ neither sendonly nor sendrecv > 100 Trying > 180 Ringing > 200 OK w/ a=sendonly > INVITE w/ a=sendrecv > 100 Trying > 200 OK w/ neither sendonly nor sendrecv > > Call that goes into on hold immediately (leg w/ device): > INVITE w/ neither, but has sdp > 407 > INVITE w/ neither, but has sdp > 100 > 183 w/ neither > 200 w/ neither > UPDATE w/ sendonly (which is after the second 200 OK above) > 200 OK w/ recvonly > > > > I'm going to try and do some more digging on this tomorrow, but if someone has any insights here on what the proper/expected behavior is, I sure would love to hear it. > > It seems like this may be somewhat related to FS-9765 - however there is no invite w/o an SDP. Also similar sounding to FS-11234. I'm happy to take this forward to provider (I'm inquiring with them as well) but would be helpful to have some more details on which piece is behaving incorrectly. > > -- Nathan > > -------- Original Message -------- > From: Nathan Neulinger > Sent: Tue, Sep 4, 2018 2:45 PM CDT > To: FreeSWITCH Users Help > > Subject: Phone spontaneously going into Hold when a call is established? > > I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside > number was going immediately to 'On Hold' when the person answered. > > Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a > polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> > 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Fri Sep 7 18:27:27 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 7 Sep 2018 13:27:27 -0500 Subject: [Freeswitch-users] Phone spontaneously going into Hold when a call is established? In-Reply-To: <72FBA931-44C1-4796-AFD0-A4956958FF1B@jerris.com> References: <72FBA931-44C1-4796-AFD0-A4956958FF1B@jerris.com> Message-ID: <85930d39-c31f-024a-5c13-002d62396c5a@mst.edu> Please note that I tracked the change in behavior to a commit related to changing SDP for video which clearly has some secondary side effects at least on this situation. See subsequent emails and FS-11385. The problem is the timing - all I know is that it's ringing, then the moment I hear audio (in my test case, when I call a number with an answering machine and it picks up) - it goes to hold and never comes of unless I toggle hold on the phone I'm calling from -- and note that it requires me to push hold key _twice_. While it starts, phone shows "Held" but I'm still hearing audio. If I then push Hold, it really goes on hold. Pushing it again, and the audio is back to normal. Is the "being answered sendonly" something that is typical? I've still got open communication with our provider, so can easily point out more if it's doing something unusual. Would it help to get a combined sip trace showing both legs in one timeline? -- Nathan ------------------------------------------------------------------------------------------------------------------------ *From:* Michael Jerris *Sent:* Fri, Sep 7, 2018 1:09 PM CDT *To:* FreeSWITCH Users Help *Subject:* [Freeswitch-users] Phone spontaneously going into Hold when a call is established? > Some notes here… if there is none of the inactive/sendonly/recvonly/sendrecv its the same as if it was sendrecv > > In the going on to on-hold immediately (leg w/ provider) case… the call is being answered sendonly, essentially they > are answering the call on hold, then sending us a re-invite to take us off hold.  Its correct that the call should > start on hold, but it should come back off hold, can you confirm that it is? > > In the going on to on-hold immediately (leg w/ device) case…  the call is being answered off hold, and then we are > getting an update with sendonly (putting us on hold)…. I would expect it to be on hold after that. > > > > >> On Sep 6, 2018, at 11:48 PM, Nathan Neulinger > wrote: >> >> Finally managed to get a reproducible failure on this, but don't understand SIP enough to know where it's going >> wrong. I have two local numbers that I can call via my local telco over a SIP trunk (MetaSwitch) - one reliably >> demonstrates the behavior, the other reliably works normal: >> >> The call that is working normally (leg w/ provider): >> INVITE w/ neither sendonly nor sendrecv >> 100 Trying >> 183 Progress w/ a=sendrecv >> 200 OK w/ a=sendrecv >> >> The call that is going into on-hold immediately (leg w/ provider): >> INVITE w/ neither sendonly nor sendrecv >> 100 Trying >> 180 Ringing >> 200 OK  w/ a=sendonly >> INVITE w/ a=sendrecv >> 100 Trying >> 200 OK w/ neither sendonly nor sendrecv >> >> Call that goes into on hold immediately (leg w/ device): >> INVITE w/ neither, but has sdp >> 407 >> INVITE w/ neither, but has sdp >> 100 >> 183 w/ neither >> 200 w/ neither >> UPDATE w/ sendonly (which is after the second 200 OK above) >> 200 OK w/ recvonly >> >> >> >> I'm going to try and do some more digging on this tomorrow, but if someone has any insights here on what the >> proper/expected behavior is, I sure would love to hear it. >> >> It seems like this may be somewhat related to FS-9765 - however there is no invite w/o an SDP. Also similar sounding >> to FS-11234. I'm happy to take this forward to provider (I'm inquiring with them as well) but would be helpful to >> have some more details on which piece is behaving incorrectly. >> >> -- Nathan >> >> -------- Original Message -------- >> From: Nathan Neulinger >> Sent: Tue, Sep 4, 2018 2:45 PM CDT >> To: FreeSWITCH Users Help > >> Subject: Phone spontaneously going into Hold when a call is established? >> >> I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside >> number was going immediately to 'On Hold' when the person answered. >> >> Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a >> polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> >> 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Sep 7 21:53:28 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 7 Sep 2018 22:53:28 +0100 Subject: [Freeswitch-users] How to use originate command call to number and then bridge to call center module In-Reply-To: References: <9af9a5f1-1f5a-bf26-d5b3-8e12b5728af7@brightfire.net> Message-ID: If i were you, I'd just use newfies dialer. Much much easier. On Fri, Sep 7, 2018, 19:33 Do Nguyen Ha wrote: > Thank you for your comments > > On Thu, Sep 6, 2018 at 12:01 AM Matthew Grooms > wrote: > >> On 9/4/2018 8:24 PM, Do Nguyen Ha wrote: >> > thank you for sharing your experience >> > >> > my goal is simple auto calling from esl to a list of numbers(callers) >> > when callers answer phone and then i will bridge to callcenter module >> > >> > to check the callers answer or not of originate command, i use the >> > event-name "BACKGROUND_JOB". the message body to check is "_body": >> > "-ERR USER_BUSY\n"/"-ERR NO_ANSWER\n"/"-ERR >> DESTINATION_OUT_OF_ORDER\n"... >> > >> >> I believe those are based on the SIP response codes. You should check >> out the commercial AMD module which will help you detect if you've >> reached a human or a machine ... >> >> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd >> >> > to check the callers is answered by the agents of callcenter i use the >> > event-name "CUSTOM" and event-name "bridge-agent-start" >> > >> > that is my dial manger logic >> > >> > is there any suggestion >> > >> >> Sounds like your on the right path. But in my experience, dialer logic >> has less to do with overcoming technical challenges and more to do with >> adhering to the legal requirements associated with outbound dialing. >> >> -Matthew >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jos at firstcom.dk Mon Sep 10 10:29:24 2018 From: jos at firstcom.dk (=?utf-8?B?Sm9uIFNjaMO4cHppbnNreQ==?=) Date: Mon, 10 Sep 2018 10:29:24 +0000 Subject: [Freeswitch-users] DTLS renegotiation Message-ID: <754A8A4C-6D94-410D-B14B-C2935ED2ABBC@firstcom.dk> Hi List, We are currently investigating a problem, where our verto based softphone client re-connects, renegotiates ICE and then fails, as DTLS is never renegotiated. We registered a ticket, FS-11268, but there has been no response yet. Anybody got an idea of where we should look for a fix? Kind Regards Jon Schøpzinsky -------------- next part -------------- An HTML attachment was scrubbed... URL: From f.antonini at tiesse.com Tue Sep 11 09:57:26 2018 From: f.antonini at tiesse.com (fabio) Date: Tue, 11 Sep 2018 11:57:26 +0200 Subject: [Freeswitch-users] how to push data frames directly to FS from an endpoint Message-ID: <3cd0fefa-84db-ccae-d8dd-476a8ad9aed5@tiesse.com> Hi All I'm developing a simple FS endpoint. At the moment I have registered a 'read_frame' function to return some data to the FS core. For what I understand this function is called by the FS core. It works as a charm. Anyway I would like to push the data frames directly to the FS core without registering a 'read_frame' function as soon as the data are available from the lower layer. Is there any FS API for this purpose? I didn't find any example about this approach. I remember that this was possible in Asterisk. Where can I found any informations about the FS internals? Any help will be greatly appreciated. Best regards Fabio -- Fabio Antonini /Software Engineer (Ph.D)/ f.antonini at tiesse.com *Tel* +39.0863.455830 *Mob* +39.393.9261941 *Fax* +39.0863.455830 Via Corradini 80 67051 Avezzano (AQ) Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in Italy. Clicca per visitare il sito Tiesse *Tiesse S.p.A.* - www.tiesse.com Via Asti 4, 10015 Ivrea (TO) Pagina Tiesse su Linkedin, clicca e visitaci *Disclaimer:* il contenuto di questa email è riservato e non vincolante per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di segnalarlo immediatamente al mittente, di non utilizzare e divulgare il contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A. declina ogni responsabilità da qualsiasi conseguenza derivante da utilizzi non autorizzati, contraffazioni o manomissioni di email recanti riferimenti all'azienda. *Rispetta l'ambiente. Non stampare questa mail se non è necessario.* -------------- next part -------------- An HTML attachment was scrubbed... URL: From ben.kaufman at altigen.com Tue Sep 11 18:33:37 2018 From: ben.kaufman at altigen.com (Ben Kaufman) Date: Tue, 11 Sep 2018 18:33:37 +0000 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI Message-ID: Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user. The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event. In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000): > REFER-TO: In the case of transfer to another Teams user, the transfer Refer-To: header looks like this: > REFER-TO: When this occurs, the last line in the Freeswitch log for the call is: > [DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com] There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report. Regards, Ben Kaufman AltiGen Communications, Inc. 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From anacelia.sarlo at gmail.com Tue Sep 11 20:51:16 2018 From: anacelia.sarlo at gmail.com (Anacelia Sarlo) Date: Tue, 11 Sep 2018 17:51:16 -0300 Subject: [Freeswitch-users] Problem with NAT & bypass media configuration Message-ID: Hi, I have the following environment : FS (ethernet interface with public address -190.64.204.60) ---- FW (provider) ------------- INTERNET ---- Router --NAT/FW----LAN Softphone (type A) INTERNET ---- Mobile network ---- Softpone (type B) Freeswitch v1.6.20, on Centos 7, behind a firewall, with sip ports tcp/udp 5560 already opened, and it is not natting. I have two type of SIP devices: Type A) connected to a LAN behind a router/firewall, natted and with upnp enabled Type B) connected through the mobile network, also natted. Calls (audio / video) between two A endpoints works!, but calls between A and B endpoints are established, but there is no audio / video, and after 30 seconds it hungup. I'am using FS for SIP signaling, and bypassing media. I tried many softphones (Linphone, antisip, and more) without luck. Also enable stun at both softphones, but I have the same result. FS is not using nat functionality (no UPNP/PMP). I have just one sip profile (external) with port 5560. I'am attaching the configuration. I suppose it's something related to RTP and NAT but I tried many configurations, and can't resolve it. Thanks a lot for your help!. Regards, Anacelia vars.xml sip_profiles/external.xml dialplan/default.xml --------------------- Sip trace (Softphone B call Softphone A) ------------ Type A softphone= portero at 190.64.204.60 Type B softphone= anacelia at 190.64.204.60 freeswitch at Vps-230738> freeswitch at Vps-230738> freeswitch at Vps-230738> freeswitch at Vps-230738> freeswitch at Vps-230738> recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:20.410606: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 Route: Max-Forwards: 70 From: "Portero" ;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47401 REGISTER User-Agent: Beward SIP Contact: Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:20.411630: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 From: "Portero" ;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" ;tag=ae7Sv6D29B5DH Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47401 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces WWW-Authenticate: Digest realm="190.64.204.60", nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:20.423845: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 Route: Max-Forwards: 70 From: "Portero" ;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47402 REGISTER User-Agent: Beward SIP Contact: Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Authorization: Digest username="portero", realm="190.64.204.60", nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", uri="sip:190.64.204.60:5560", response="f5a8e2e4021c06ed681121f5a0deaf1d", algorithm=MD5, cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:20.426835: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 From: "Portero" ;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" ;tag=BQ0jy1y56mU0c Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47402 REGISTER Contact: ;expires=30 Date: Tue, 11 Sep 2018 20:31:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 771 bytes to udp/[167.61.200.85]:5560 at 17:31:20.488658: ------------------------------------------------------------------------ NOTIFY sip:portero at 167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bKZcm1vQ4N7ajyN Max-Forwards: 70 From: ;tag=c0SB0vF93XHKr To: Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a CSeq: 128012876 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: message-summary Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 68 Messages-Waiting: no Message-Account: sip:portero at 190.64.204.60 ------------------------------------------------------------------------ recv 319 bytes from udp/[167.61.200.85]:5560 at 17:31:20.832843: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bKZcm1vQ4N7ajyN Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a From: ;tag=c0SB0vF93XHKr To: ;tag=z9hG4bKZcm1vQ4N7ajyN CSeq: 128012876 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ recv 1346 bytes from udp/[186.51.236.123]:38474 at 17:31:21.388899: ------------------------------------------------------------------------ INVITE sip:portero at 190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.tPc8Rktcw;rport From: ;tag=v0lM8TzLu To: sip:portero at 190.64.204.60 CSeq: 20 INVITE Call-ID: thpKqQo3Kf Max-Forwards: 70 Supported: replaces, outbound, gruu Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 496 Contact: ;+sip.instance="";+org.linphone.specs=groupchat User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) v=0 o=anacelia 1261 1974 IN IP4 10.45.210.97 s=Talk c=IN IP4 10.45.210.97 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:97 speex/16000 a=fmtp:97 vbr=on a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:99 telephone-event/16000 a=rtpmap:100 telephone-event/8000 a=rtcp-fb:* ccm tmmbr ------------------------------------------------------------------------ send 305 bytes to udp/[186.51.236.123]:38474 at 17:31:21.389159: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: ;tag=v0lM8TzLu To: sip:portero at 190.64.204.60 Call-ID: thpKqQo3Kf CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel sofia/external/anacelia at 190.64.204.60 [a4fa5a3c-b601-11e8-9842-cdce49bc5d42] 2018-09-11 17:31:21.377624 [INFO] mod_dialplan_xml.c:637 Processing anacelia ->portero in context default 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel sofia/external/portero at 167.61.200.85:5560 [a4fab0e0-b601-11e8-984c-cdce49bc5d42] send 1292 bytes to udp/[167.61.200.85]:5560 at 17:31:21.392377: ------------------------------------------------------------------------ INVITE sip:portero at 167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK0NDtyjNS4K8gH Max-Forwards: 69 From: "anacelia" ;tag=ejcX3jHgyFyrF To: Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012876 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 496 X-FS-Support: update_display,send_info Remote-Party-ID: "anacelia" ;party=calling;screen=yes;privacy=off v=0 o=anacelia 1261 1974 IN IP4 10.45.210.97 s=Talk c=IN IP4 10.45.210.97 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:97 speex/16000 a=fmtp:97 vbr=on a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:99 telephone-event/16000 a=rtpmap:100 telephone-event/8000 a=rtcp-fb:* ccm tmmbr ------------------------------------------------------------------------ recv 313 bytes from udp/[167.61.200.85]:5560 at 17:31:21.447821: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" ;tag=ejcX3jHgyFyrF To: CSeq: 128012876 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 453 bytes from udp/[167.61.200.85]:5560 at 17:31:21.466797: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" ;tag=ejcX3jHgyFyrF To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012876 INVITE Contact: Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:21.457617 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/ portero at 167.61.200.85:5560! 2018-09-11 17:31:21.457617 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/anacelia at 190.64.204.60! send 675 bytes to udp/[186.51.236.123]:38474 at 17:31:21.473235: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: ;tag=v0lM8TzLu To: ;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 20 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2018-09-11 17:31:21.457617 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/anacelia at 190.64.204.60! recv 817 bytes from udp/[167.61.200.85]:5560 at 17:31:22.018913: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" ;tag=ejcX3jHgyFyrF To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012876 INVITE Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Contact: Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 287 v=0 o=- 3745675951 3745675952 IN IP4 192.168.1.139 s=Door station call b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 100 c=IN IP4 192.168.1.139 b=TIAS:64000 a=rtcp:4003 IN IP4 192.168.1.139 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 ------------------------------------------------------------------------ send 413 bytes to udp/[167.61.200.85]:5560 at 17:31:22.019799: ------------------------------------------------------------------------ ACK sip:portero at 167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK1y6j0D6v1vy3c Max-Forwards: 70 From: "anacelia" ;tag=ejcX3jHgyFyrF To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012876 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:22.017623 [NOTICE] sofia.c:8180 Channel [sofia/external/ portero at 167.61.200.85:5560] has been answered 2018-09-11 17:31:22.017623 [NOTICE] switch_ivr.c:779 Channel [sofia/external/anacelia at 190.64.204.60] has been answered send 983 bytes to udp/[186.51.236.123]:38474 at 17:31:22.035733: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: ;tag=v0lM8TzLu To: ;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 20 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 275 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=- 3745675951 3745675952 IN IP4 192.168.1.139 s=Door station call b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 100 c=IN IP4 192.168.1.139 b=TIAS:64000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=rtcp:4003 IN IP4 192.168.1.139 ------------------------------------------------------------------------ recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:22.426439: ------------------------------------------------------------------------ ACK sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;rport;branch=z9hG4bK.bkN7s~~XL From: ;tag=v0lM8TzLu To: ;tag=D9j41Q0c1675K CSeq: 20 ACK Call-ID: thpKqQo3Kf Max-Forwards: 70 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) ------------------------------------------------------------------------ recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:45.458386: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry Route: Max-Forwards: 70 From: "Portero" ;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47403 REGISTER User-Agent: Beward SIP Contact: Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:45.459222: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry From: "Portero" ;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" ;tag=FU5N5D2KUrmBB Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47403 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces WWW-Authenticate: Digest realm="190.64.204.60", nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:45.477423: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w Route: Max-Forwards: 70 From: "Portero" ;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47404 REGISTER User-Agent: Beward SIP Contact: Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Authorization: Digest username="portero", realm="190.64.204.60", nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", uri="sip:190.64.204.60:5560", response="01447ae196414735cbe9fa0da2048d18", algorithm=MD5, cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:45.479598: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w From: "Portero" ;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" ;tag=g4ye78jQr1ayp Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47404 REGISTER Contact: ;expires=30 Date: Tue, 11 Sep 2018 20:31:45 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:53.066351: ------------------------------------------------------------------------ BYE sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.MOC0o5vx8;rport From: ;tag=v0lM8TzLu To: ;tag=D9j41Q0c1675K CSeq: 21 BYE Call-ID: thpKqQo3Kf Max-Forwards: 70 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] sofia.c:1012 Hangup sofia/external/ anacelia at 190.64.204.60 [CS_HIBERNATE] [NORMAL_CLEARING] send 442 bytes to udp/[186.51.236.123]:38474 at 17:31:53.067386: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.MOC0o5vx8;rport=38474;received=186.51.236.123 From: ;tag=v0lM8TzLu To: ;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 21 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] switch_ivr_bridge.c:1302 Hangup sofia/external/portero at 167.61.200.85:5560 [CS_HIBERNATE] [NORMAL_CLEARING] send 586 bytes to udp/[167.61.200.85]:5560 at 17:31:53.068575: ------------------------------------------------------------------------ BYE sip:portero at 167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK27ZB28p0y5mpr Max-Forwards: 70 From: "anacelia" ;tag=ejcX3jHgyFyrF To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012877 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 3 (sofia/external/anacelia at 190.64.204.60) Ended 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/anacelia at 190.64.204.60 [CS_DESTROY] 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 4 (sofia/external/portero at 167.61.200.85:5560) Ended 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/portero at 167.61.200.85:5560 [CS_DESTROY] recv 343 bytes from udp/[167.61.200.85]:5560 at 17:31:53.083598: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK27ZB28p0y5mpr Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" ;tag=ejcX3jHgyFyrF To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012877 BYE Content-Length: 0 ------------------------------------------------------------------------ recv 1062 bytes from udp/[186.51.236.123]:38474 at 17:31:53.537993: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.FMbxN6bPS;rport From: ;tag=MBKS11QUc To: sip:anacelia at 190.64.204.60 CSeq: 24 REGISTER Call-ID: DFvn85sO3E Max-Forwards: 70 Supported: replaces, outbound, gruu Accept: application/sdp Accept: text/plain Accept: application/vnd.gsma.rcs-ft-http+xml Contact: ;+sip.instance="";+org.linphone.specs=groupchat Expires: 3600 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) Authorization: Digest realm="190.64.204.60", nonce="6fce67e0-b601-11e8-982b-cdce49bc5d42", algorithm=MD5, username="anacelia", uri="sip:190.64.204.60:5560", response="077e0b6c433066db828f9ec829f8d558", cnonce="~Oh2sj~-KZHHDh9I", nc=00000004, qop=auth ------------------------------------------------------------------------ send 765 bytes to udp/[186.51.236.123]:38474 at 17:31:53.540658: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.FMbxN6bPS;rport=38474;received=186.51.236.123 From: ;tag=MBKS11QUc To: ;tag=HDr7833tNa1gj Call-ID: DFvn85sO3E CSeq: 24 REGISTER Contact: ;expires=30 Date: Tue, 11 Sep 2018 20:31:53 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Wed Sep 12 03:56:56 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 12 Sep 2018 12:56:56 +0900 Subject: [Freeswitch-users] how to push data frames directly to FS from an endpoint In-Reply-To: <3cd0fefa-84db-ccae-d8dd-476a8ad9aed5@tiesse.com> References: <3cd0fefa-84db-ccae-d8dd-476a8ad9aed5@tiesse.com> Message-ID: Hi, Try this one https://docs.freeswitch.org/ Also some awesome FreeSWITCH books are around. Try Mastering FreeSWITCH, it has a chapter on module development. Regards, Alex On Wed, Sep 12, 2018 at 2:55 AM fabio wrote: > Hi All > > I'm developing a simple FS endpoint. At the moment I have registered a > 'read_frame' function to return some data to the FS core. > > For what I understand this function is called by the FS core. It works as > a charm. > > > Anyway I would like to push the data frames directly to the FS core > without registering a 'read_frame' function as soon as the data are > available from the lower layer. > > Is there any FS API for this purpose? I didn't find any example about this > approach. > > I remember that this was possible in Asterisk. > > > Where can I found any informations about the FS internals? > > > Any help will be greatly appreciated. > > Best regards > > Fabio > -- > > Fabio Antonini > > *Software Engineer (Ph.D)* > f.antonini at tiesse.com > *Tel* +39.0863.455830 > *Mob* +39.393.9261941 > *Fax* +39.0863.455830 > Via Corradini 80 > 67051 Avezzano (AQ) > > [image: Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in > Italy. Clicca per visitare il sito Tiesse] > > *Tiesse S.p.A.* - www.tiesse.com > Via Asti 4, 10015 Ivrea (TO) > [image: Pagina Tiesse su Linkedin, clicca e visitaci] > > *Disclaimer:* il contenuto di questa email è riservato e non vincolante > per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di > segnalarlo immediatamente al mittente, di non utilizzare e divulgare il > contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A. > declina ogni responsabilità da qualsiasi conseguenza derivante da utilizzi > non autorizzati, contraffazioni o manomissioni di email recanti riferimenti > all'azienda. > *Rispetta l'ambiente. Non stampare questa mail se non è necessario.* > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ko at sv01.de Wed Sep 12 06:41:15 2018 From: ko at sv01.de (Kevin Olbrich) Date: Wed, 12 Sep 2018 08:41:15 +0200 Subject: [Freeswitch-users] Problem with NAT & bypass media configuration In-Reply-To: References: Message-ID: Hi! It realy sounds like you want to set bypass-media to false. > When set, the media (RTP) from the originating endpoint is sent directly to the destination endpoint and vice versa. The signaling (SIP) for both endpoints still goes through FreeSWITCH, but the media is point-to-point. If set to true like in your config, media is flowing from the endpoint to the other endpoint which is much more harder for any firewall or NAT in it's way. Also many carriers block RTP flows from IPs other than the SIP registration, which also results in the same problem. Kind regards Kevin Am Di., 11. Sep. 2018 um 23:56 Uhr schrieb Anacelia Sarlo < anacelia.sarlo at gmail.com>: > Hi, I have the following environment : > > > FS (ethernet interface with public address -190.64.204.60) ---- FW > (provider) ------------- INTERNET ---- Router --NAT/FW----LAN Softphone > (type A) > INTERNET ---- Mobile network ---- > Softpone (type B) > Freeswitch v1.6.20, on Centos 7, behind a firewall, with sip ports tcp/udp > 5560 already opened, and it is not natting. > I have two type of SIP devices: > Type A) connected to a LAN behind a router/firewall, natted and with upnp > enabled > Type B) connected through the mobile network, also natted. > > Calls (audio / video) between two A endpoints works!, but calls between A > and B endpoints are established, but there is no audio / video, and after > 30 seconds it hungup. I'am using FS for SIP signaling, and bypassing media. > I tried many softphones (Linphone, antisip, and more) without luck. Also > enable stun at both softphones, but I have the same result. > > FS is not using nat functionality (no UPNP/PMP). > I have just one sip profile (external) with port 5560. > > I'am attaching the configuration. I suppose it's something related to RTP > and NAT but I tried many configurations, and can't resolve it. > Thanks a lot for your help!. > Regards, > Anacelia > > > vars.xml > > > > data="sound_prefix=$${sounds_dir}/en/us/callie"/> > > > > > data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/> > > > > > > > > > > > > > > > data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/> > > > > > > > > > > > > > data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> > > > data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> > > > > > > > > > > data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> > > data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> > > data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> > data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/> > data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/> > data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/> > > > > > > > > data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/> > > > > > > > > > > > > > data="video_mute_png=$${images_dir}/default-mute.png"/> > data="video_no_avatar_png=$${images_dir}/default-avatar.png"/> > > > > sip_profiles/external.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > dialplan/default.xml > > > > > > > > data="nolocal:absolute_codec_string=${ep_codec_string}"/> > > > > > > > > > > > data="absolute_codec_string=PCMU,PCMA,H264"/> > > > > > > > > > > --------------------- > > Sip trace (Softphone B call Softphone A) > ------------ > Type A softphone= portero at 190.64.204.60 > Type B softphone= anacelia at 190.64.204.60 > > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> recv 539 bytes from udp/[167.61.200.85]:5560 at > 17:31:20.410606: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 > Route: > Max-Forwards: 70 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47401 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:20.411630: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" ;tag=ae7Sv6D29B5DH > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47401 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="190.64.204.60", > nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:20.423845: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 > Route: > Max-Forwards: 70 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47402 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Authorization: Digest username="portero", realm="190.64.204.60", > nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", uri="sip:190.64.204.60:5560", > response="f5a8e2e4021c06ed681121f5a0deaf1d", algorithm=MD5, > cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:20.426835: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" ;tag=BQ0jy1y56mU0c > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47402 REGISTER > Contact: ;expires=30 > Date: Tue, 11 Sep 2018 20:31:20 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 771 bytes to udp/[167.61.200.85]:5560 at 17:31:20.488658: > ------------------------------------------------------------------------ > NOTIFY sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bKZcm1vQ4N7ajyN > Max-Forwards: 70 > From: ;tag=c0SB0vF93XHKr > To: > Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a > CSeq: 128012876 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Event: message-summary > Allow-Events: talk, hold, conference, refer > Subscription-State: terminated;reason=noresource > Content-Type: application/simple-message-summary > Content-Length: 68 > > Messages-Waiting: no > Message-Account: sip:portero at 190.64.204.60 > > ------------------------------------------------------------------------ > recv 319 bytes from udp/[167.61.200.85]:5560 at 17:31:20.832843: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bKZcm1vQ4N7ajyN > Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a > From: ;tag=c0SB0vF93XHKr > To: ;tag=z9hG4bKZcm1vQ4N7ajyN > CSeq: 128012876 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1346 bytes from udp/[186.51.236.123]:38474 at 17:31:21.388899: > ------------------------------------------------------------------------ > INVITE sip:portero at 190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.tPc8Rktcw;rport > From: ;tag=v0lM8TzLu > To: sip:portero at 190.64.204.60 > CSeq: 20 INVITE > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > Supported: replaces, outbound, gruu > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO, UPDATE > Content-Type: application/sdp > Content-Length: 496 > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="";+org.linphone.specs=groupchat > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > v=0 > o=anacelia 1261 1974 IN IP4 10.45.210.97 > s=Talk > c=IN IP4 10.45.210.97 > t=0 0 > a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics > m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 > a=rtpmap:96 opus/48000/2 > a=fmtp:96 useinbandfec=1 > a=rtpmap:97 speex/16000 > a=fmtp:97 vbr=on > a=rtpmap:98 speex/8000 > a=fmtp:98 vbr=on > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/48000 > a=rtpmap:99 telephone-event/16000 > a=rtpmap:100 telephone-event/8000 > a=rtcp-fb:* ccm tmmbr > ------------------------------------------------------------------------ > send 305 bytes to udp/[186.51.236.123]:38474 at 17:31:21.389159: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: sip:portero at 190.64.204.60 > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/anacelia at 190.64.204.60 > [a4fa5a3c-b601-11e8-9842-cdce49bc5d42] > 2018-09-11 17:31:21.377624 [INFO] mod_dialplan_xml.c:637 Processing > anacelia ->portero in context default > 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/portero at 167.61.200.85:5560 > [a4fab0e0-b601-11e8-984c-cdce49bc5d42] > send 1292 bytes to udp/[167.61.200.85]:5560 at 17:31:21.392377: > ------------------------------------------------------------------------ > INVITE sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK0NDtyjNS4K8gH > Max-Forwards: 69 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012876 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 496 > X-FS-Support: update_display,send_info > Remote-Party-ID: "anacelia" >;party=calling;screen=yes;privacy=off > > v=0 > o=anacelia 1261 1974 IN IP4 10.45.210.97 > s=Talk > c=IN IP4 10.45.210.97 > t=0 0 > a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics > m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 > a=rtpmap:96 opus/48000/2 > a=fmtp:96 useinbandfec=1 > a=rtpmap:97 speex/16000 > a=fmtp:97 vbr=on > a=rtpmap:98 speex/8000 > a=fmtp:98 vbr=on > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/48000 > a=rtpmap:99 telephone-event/16000 > a=rtpmap:100 telephone-event/8000 > a=rtcp-fb:* ccm tmmbr > ------------------------------------------------------------------------ > recv 313 bytes from udp/[167.61.200.85]:5560 at 17:31:21.447821: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: > CSeq: 128012876 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[167.61.200.85]:5560 at 17:31:21.466797: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012876 INVITE > Contact: > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.457617 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/ > portero at 167.61.200.85:5560! > 2018-09-11 17:31:21.457617 [NOTICE] mod_sofia.c:2273 Ring-Ready > sofia/external/anacelia at 190.64.204.60! > send 675 bytes to udp/[186.51.236.123]:38474 at 17:31:21.473235: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Length: 0 > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.457617 [NOTICE] switch_ivr_originate.c:525 Ring Ready > sofia/external/anacelia at 190.64.204.60! > recv 817 bytes from udp/[167.61.200.85]:5560 at 17:31:22.018913: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012876 INVITE > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Contact: > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length: 287 > > v=0 > o=- 3745675951 3745675952 IN IP4 192.168.1.139 > s=Door station call > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 0 100 > c=IN IP4 192.168.1.139 > b=TIAS:64000 > a=rtcp:4003 IN IP4 192.168.1.139 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 telephone-event/8000 > a=fmtp:100 0-16 > ------------------------------------------------------------------------ > send 413 bytes to udp/[167.61.200.85]:5560 at 17:31:22.019799: > ------------------------------------------------------------------------ > ACK sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK1y6j0D6v1vy3c > Max-Forwards: 70 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012876 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:22.017623 [NOTICE] sofia.c:8180 Channel [sofia/external/ > portero at 167.61.200.85:5560] has been answered > 2018-09-11 17:31:22.017623 [NOTICE] switch_ivr.c:779 Channel > [sofia/external/anacelia at 190.64.204.60] has been answered > send 983 bytes to udp/[186.51.236.123]:38474 at 17:31:22.035733: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 275 > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no > > v=0 > o=- 3745675951 3745675952 IN IP4 192.168.1.139 > s=Door station call > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 0 100 > c=IN IP4 192.168.1.139 > b=TIAS:64000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 telephone-event/8000 > a=fmtp:100 0-16 > a=rtcp:4003 IN IP4 192.168.1.139 > ------------------------------------------------------------------------ > recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:22.426439: > ------------------------------------------------------------------------ > ACK sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;rport;branch=z9hG4bK.bkN7s~~XL > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > CSeq: 20 ACK > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > ------------------------------------------------------------------------ > recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:45.458386: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry > Route: > Max-Forwards: 70 > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47403 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:45.459222: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" ;tag=FU5N5D2KUrmBB > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47403 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="190.64.204.60", > nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:45.477423: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w > Route: > Max-Forwards: 70 > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47404 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Authorization: Digest username="portero", realm="190.64.204.60", > nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", uri="sip:190.64.204.60:5560", > response="01447ae196414735cbe9fa0da2048d18", algorithm=MD5, > cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:45.479598: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" ;tag=g4ye78jQr1ayp > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47404 REGISTER > Contact: ;expires=30 > Date: Tue, 11 Sep 2018 20:31:45 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:53.066351: > ------------------------------------------------------------------------ > BYE sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.MOC0o5vx8;rport > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > CSeq: 21 BYE > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] sofia.c:1012 Hangup sofia/external/ > anacelia at 190.64.204.60 [CS_HIBERNATE] [NORMAL_CLEARING] > send 442 bytes to udp/[186.51.236.123]:38474 at 17:31:53.067386: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.MOC0o5vx8;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 21 BYE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] switch_ivr_bridge.c:1302 Hangup > sofia/external/portero at 167.61.200.85:5560 [CS_HIBERNATE] [NORMAL_CLEARING] > send 586 bytes to udp/[167.61.200.85]:5560 at 17:31:53.068575: > ------------------------------------------------------------------------ > BYE sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK27ZB28p0y5mpr > Max-Forwards: 70 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012877 BYE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Reason: Q.850;cause=16;text="NORMAL_CLEARING" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 3 > (sofia/external/anacelia at 190.64.204.60) Ended > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close > Channel sofia/external/anacelia at 190.64.204.60 [CS_DESTROY] > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 4 > (sofia/external/portero at 167.61.200.85:5560) Ended > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close > Channel sofia/external/portero at 167.61.200.85:5560 [CS_DESTROY] > recv 343 bytes from udp/[167.61.200.85]:5560 at 17:31:53.083598: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK27ZB28p0y5mpr > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012877 BYE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1062 bytes from udp/[186.51.236.123]:38474 at 17:31:53.537993: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.FMbxN6bPS;rport > From: ;tag=MBKS11QUc > To: sip:anacelia at 190.64.204.60 > CSeq: 24 REGISTER > Call-ID: DFvn85sO3E > Max-Forwards: 70 > Supported: replaces, outbound, gruu > Accept: application/sdp > Accept: text/plain > Accept: application/vnd.gsma.rcs-ft-http+xml > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="";+org.linphone.specs=groupchat > Expires: 3600 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > Authorization: Digest realm="190.64.204.60", > nonce="6fce67e0-b601-11e8-982b-cdce49bc5d42", algorithm=MD5, > username="anacelia", uri="sip:190.64.204.60:5560", > response="077e0b6c433066db828f9ec829f8d558", cnonce="~Oh2sj~-KZHHDh9I", > nc=00000004, qop=auth > > ------------------------------------------------------------------------ > send 765 bytes to udp/[186.51.236.123]:38474 at 17:31:53.540658: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.FMbxN6bPS;rport=38474;received=186.51.236.123 > From: ;tag=MBKS11QUc > To: ;tag=HDr7833tNa1gj > Call-ID: DFvn85sO3E > CSeq: 24 REGISTER > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;expires=30 > Date: Tue, 11 Sep 2018 20:31:53 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jaan.Kaja at enghouse.com Wed Sep 12 07:38:25 2018 From: Jaan.Kaja at enghouse.com (Jaan Kaja) Date: Wed, 12 Sep 2018 07:38:25 +0000 Subject: [Freeswitch-users] No ringtone to caller when third party call control (3PCC) is enabled Message-ID: <8d946ef6bbaa47319a2712a8270d3cee@enghouse.com> Hi, I have configured "enable-3pcc=proxy" in the SIP profile, and have an external application that controls answer-call, make-call, etc. The configuration means that incoming calls w/o SDP are accepted, which works fine. The problem is that when I make an outgoing call, ringtone is not played to the caller, although the "ringback" variable is set for the caller's leg. I have checked the code in mod_sofia, and in sofia.c, there is a piece of code: if (sofia_test_flag(tech_pvt, TFLAG_SKIP_EARLY)) { sofia_clear_flag_locked(tech_pvt, TFLAG_SKIP_EARLY); goto done; } which disables reaching the ring-ready state. TFLAG_SKIP_EARLY is set when TFLAG_3PCC is true. If I comment out the goto statement, the A-party gets ringtone. N.B.: This is just a minimal quick fix, not a proposed solution, which works because I'm not doing an enterprise originate. Question: Is there a reason for this behavior? Documentation shows that early media must be disabled when an enterprise originate is done, but why is playing ringtone disabled in this case? Should I submit a bug report to Jira? Best regards, Jaan -------------- next part -------------- An HTML attachment was scrubbed... URL: From sukithaj at gmail.com Wed Sep 12 07:52:33 2018 From: sukithaj at gmail.com (sukitha jayasinghe) Date: Wed, 12 Sep 2018 13:22:33 +0530 Subject: [Freeswitch-users] Freeswitch 1.8.1 build without core pgsql support Message-ID: Dear All, I installed freeswitch 1.8.1 in Debian 9 using apt commands. It seems like packages has build without pgsql support. When I run the application with pgsql configurations it gives bellow error. switch_core_sqldb.c:468 Failure! PGSQL NOT AVAILABLE! is there any special reason to remove this from prebuild packages. Is there anyway to enable pgsql support without rebuild. BR, Sukitha. -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Wed Sep 12 09:26:29 2018 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 12 Sep 2018 10:26:29 +0100 Subject: [Freeswitch-users] Problem with NAT & bypass media configuration In-Reply-To: References: Message-ID: The endpoints need to send the packets directly to each other for bypass_media to work. The firewalls and/or NAT in place will be preventing that from working. Even if they know each other's public IP and port to send RTP packets to the firewall might only have opened that port up to the FS host. There are also different flavours of NAT. In some the mapping between the internal and external port numbers will be for a specific sending host and might be different for another host. So the external RTP port for the A endpoint may be valid for the FS host to send to but invalid for the B endpoint to send to, and visa versa. In short if the endpoints are on different networks bypass_media is unlikely to work reliably. On Tue, 11 Sep 2018 at 23:46, Anacelia Sarlo wrote: > Hi, I have the following environment : > > > FS (ethernet interface with public address -190.64.204.60) ---- FW > (provider) ------------- INTERNET ---- Router --NAT/FW----LAN Softphone > (type A) > INTERNET ---- Mobile network ---- > Softpone (type B) > Freeswitch v1.6.20, on Centos 7, behind a firewall, with sip ports tcp/udp > 5560 already opened, and it is not natting. > I have two type of SIP devices: > Type A) connected to a LAN behind a router/firewall, natted and with upnp > enabled > Type B) connected through the mobile network, also natted. > > Calls (audio / video) between two A endpoints works!, but calls between A > and B endpoints are established, but there is no audio / video, and after > 30 seconds it hungup. I'am using FS for SIP signaling, and bypassing media. > I tried many softphones (Linphone, antisip, and more) without luck. Also > enable stun at both softphones, but I have the same result. > > FS is not using nat functionality (no UPNP/PMP). > I have just one sip profile (external) with port 5560. > > I'am attaching the configuration. I suppose it's something related to RTP > and NAT but I tried many configurations, and can't resolve it. > Thanks a lot for your help!. > Regards, > Anacelia > > > vars.xml > > > > data="sound_prefix=$${sounds_dir}/en/us/callie"/> > > > > > data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/> > > > > > > > > > > > > > > > data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/> > > > > > > > > > > > > > data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> > > > data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> > > > > > > > > > > data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> > > data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> > > data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> > data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/> > data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/> > data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/> > > > > > > > > data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/> > > > > > > > > > > > > > data="video_mute_png=$${images_dir}/default-mute.png"/> > data="video_no_avatar_png=$${images_dir}/default-avatar.png"/> > > > > sip_profiles/external.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > dialplan/default.xml > > > > > > > > data="nolocal:absolute_codec_string=${ep_codec_string}"/> > > > > > > > > > > > data="absolute_codec_string=PCMU,PCMA,H264"/> > > > > > > > > > > --------------------- > > Sip trace (Softphone B call Softphone A) > ------------ > Type A softphone= portero at 190.64.204.60 > Type B softphone= anacelia at 190.64.204.60 > > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> > freeswitch at Vps-230738> recv 539 bytes from udp/[167.61.200.85]:5560 at > 17:31:20.410606: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 > Route: > Max-Forwards: 70 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47401 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:20.411630: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" ;tag=ae7Sv6D29B5DH > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47401 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="190.64.204.60", > nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:20.423845: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 > Route: > Max-Forwards: 70 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47402 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Authorization: Digest username="portero", realm="190.64.204.60", > nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", uri="sip:190.64.204.60:5560", > response="f5a8e2e4021c06ed681121f5a0deaf1d", algorithm=MD5, > cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:20.426835: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 > From: "Portero" >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA > To: "Portero" ;tag=BQ0jy1y56mU0c > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47402 REGISTER > Contact: ;expires=30 > Date: Tue, 11 Sep 2018 20:31:20 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 771 bytes to udp/[167.61.200.85]:5560 at 17:31:20.488658: > ------------------------------------------------------------------------ > NOTIFY sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bKZcm1vQ4N7ajyN > Max-Forwards: 70 > From: ;tag=c0SB0vF93XHKr > To: > Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a > CSeq: 128012876 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Event: message-summary > Allow-Events: talk, hold, conference, refer > Subscription-State: terminated;reason=noresource > Content-Type: application/simple-message-summary > Content-Length: 68 > > Messages-Waiting: no > Message-Account: sip:portero at 190.64.204.60 > > ------------------------------------------------------------------------ > recv 319 bytes from udp/[167.61.200.85]:5560 at 17:31:20.832843: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bKZcm1vQ4N7ajyN > Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a > From: ;tag=c0SB0vF93XHKr > To: ;tag=z9hG4bKZcm1vQ4N7ajyN > CSeq: 128012876 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1346 bytes from udp/[186.51.236.123]:38474 at 17:31:21.388899: > ------------------------------------------------------------------------ > INVITE sip:portero at 190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.tPc8Rktcw;rport > From: ;tag=v0lM8TzLu > To: sip:portero at 190.64.204.60 > CSeq: 20 INVITE > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > Supported: replaces, outbound, gruu > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO, UPDATE > Content-Type: application/sdp > Content-Length: 496 > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="";+org.linphone.specs=groupchat > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > v=0 > o=anacelia 1261 1974 IN IP4 10.45.210.97 > s=Talk > c=IN IP4 10.45.210.97 > t=0 0 > a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics > m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 > a=rtpmap:96 opus/48000/2 > a=fmtp:96 useinbandfec=1 > a=rtpmap:97 speex/16000 > a=fmtp:97 vbr=on > a=rtpmap:98 speex/8000 > a=fmtp:98 vbr=on > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/48000 > a=rtpmap:99 telephone-event/16000 > a=rtpmap:100 telephone-event/8000 > a=rtcp-fb:* ccm tmmbr > ------------------------------------------------------------------------ > send 305 bytes to udp/[186.51.236.123]:38474 at 17:31:21.389159: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: sip:portero at 190.64.204.60 > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/anacelia at 190.64.204.60 > [a4fa5a3c-b601-11e8-9842-cdce49bc5d42] > 2018-09-11 17:31:21.377624 [INFO] mod_dialplan_xml.c:637 Processing > anacelia ->portero in context default > 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/portero at 167.61.200.85:5560 > [a4fab0e0-b601-11e8-984c-cdce49bc5d42] > send 1292 bytes to udp/[167.61.200.85]:5560 at 17:31:21.392377: > ------------------------------------------------------------------------ > INVITE sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK0NDtyjNS4K8gH > Max-Forwards: 69 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012876 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 496 > X-FS-Support: update_display,send_info > Remote-Party-ID: "anacelia" >;party=calling;screen=yes;privacy=off > > v=0 > o=anacelia 1261 1974 IN IP4 10.45.210.97 > s=Talk > c=IN IP4 10.45.210.97 > t=0 0 > a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics > m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 > a=rtpmap:96 opus/48000/2 > a=fmtp:96 useinbandfec=1 > a=rtpmap:97 speex/16000 > a=fmtp:97 vbr=on > a=rtpmap:98 speex/8000 > a=fmtp:98 vbr=on > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/48000 > a=rtpmap:99 telephone-event/16000 > a=rtpmap:100 telephone-event/8000 > a=rtcp-fb:* ccm tmmbr > ------------------------------------------------------------------------ > recv 313 bytes from udp/[167.61.200.85]:5560 at 17:31:21.447821: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: > CSeq: 128012876 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[167.61.200.85]:5560 at 17:31:21.466797: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012876 INVITE > Contact: > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.457617 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/ > portero at 167.61.200.85:5560! > 2018-09-11 17:31:21.457617 [NOTICE] mod_sofia.c:2273 Ring-Ready > sofia/external/anacelia at 190.64.204.60! > send 675 bytes to udp/[186.51.236.123]:38474 at 17:31:21.473235: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Length: 0 > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2018-09-11 17:31:21.457617 [NOTICE] switch_ivr_originate.c:525 Ring Ready > sofia/external/anacelia at 190.64.204.60! > recv 817 bytes from udp/[167.61.200.85]:5560 at 17:31:22.018913: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012876 INVITE > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Contact: > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length: 287 > > v=0 > o=- 3745675951 3745675952 IN IP4 192.168.1.139 > s=Door station call > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 0 100 > c=IN IP4 192.168.1.139 > b=TIAS:64000 > a=rtcp:4003 IN IP4 192.168.1.139 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 telephone-event/8000 > a=fmtp:100 0-16 > ------------------------------------------------------------------------ > send 413 bytes to udp/[167.61.200.85]:5560 at 17:31:22.019799: > ------------------------------------------------------------------------ > ACK sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK1y6j0D6v1vy3c > Max-Forwards: 70 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012876 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:22.017623 [NOTICE] sofia.c:8180 Channel [sofia/external/ > portero at 167.61.200.85:5560] has been answered > 2018-09-11 17:31:22.017623 [NOTICE] switch_ivr.c:779 Channel > [sofia/external/anacelia at 190.64.204.60] has been answered > send 983 bytes to udp/[186.51.236.123]:38474 at 17:31:22.035733: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 20 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 275 > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no > > v=0 > o=- 3745675951 3745675952 IN IP4 192.168.1.139 > s=Door station call > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 0 100 > c=IN IP4 192.168.1.139 > b=TIAS:64000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 telephone-event/8000 > a=fmtp:100 0-16 > a=rtcp:4003 IN IP4 192.168.1.139 > ------------------------------------------------------------------------ > recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:22.426439: > ------------------------------------------------------------------------ > ACK sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;rport;branch=z9hG4bK.bkN7s~~XL > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > CSeq: 20 ACK > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > ------------------------------------------------------------------------ > recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:45.458386: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry > Route: > Max-Forwards: 70 > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47403 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:45.459222: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" ;tag=FU5N5D2KUrmBB > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47403 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > WWW-Authenticate: Digest realm="190.64.204.60", > nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:45.477423: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w > Route: > Max-Forwards: 70 > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47404 REGISTER > User-Agent: Beward SIP > Contact: > Expires: 300 > Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS > Authorization: Digest username="portero", realm="190.64.204.60", > nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", uri="sip:190.64.204.60:5560", > response="01447ae196414735cbe9fa0da2048d18", algorithm=MD5, > cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:45.479598: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 167.61.200.85:5560 > ;rport=5560;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w > From: "Portero" >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 > To: "Portero" ;tag=g4ye78jQr1ayp > Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE > CSeq: 47404 REGISTER > Contact: ;expires=30 > Date: Tue, 11 Sep 2018 20:31:45 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:53.066351: > ------------------------------------------------------------------------ > BYE sip:portero at 190.64.204.60:5560;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.MOC0o5vx8;rport > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > CSeq: 21 BYE > Call-ID: thpKqQo3Kf > Max-Forwards: 70 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] sofia.c:1012 Hangup sofia/external/ > anacelia at 190.64.204.60 [CS_HIBERNATE] [NORMAL_CLEARING] > send 442 bytes to udp/[186.51.236.123]:38474 at 17:31:53.067386: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.MOC0o5vx8;rport=38474;received=186.51.236.123 > From: ;tag=v0lM8TzLu > To: ;tag=D9j41Q0c1675K > Call-ID: thpKqQo3Kf > CSeq: 21 BYE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] switch_ivr_bridge.c:1302 Hangup > sofia/external/portero at 167.61.200.85:5560 [CS_HIBERNATE] [NORMAL_CLEARING] > send 586 bytes to udp/[167.61.200.85]:5560 at 17:31:53.068575: > ------------------------------------------------------------------------ > BYE sip:portero at 167.61.200.85:5560;ob SIP/2.0 > Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK27ZB28p0y5mpr > Max-Forwards: 70 > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > CSeq: 128012877 BYE > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Reason: Q.850;cause=16;text="NORMAL_CLEARING" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 3 > (sofia/external/anacelia at 190.64.204.60) Ended > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close > Channel sofia/external/anacelia at 190.64.204.60 [CS_DESTROY] > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 4 > (sofia/external/portero at 167.61.200.85:5560) Ended > 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close > Channel sofia/external/portero at 167.61.200.85:5560 [CS_DESTROY] > recv 343 bytes from udp/[167.61.200.85]:5560 at 17:31:53.083598: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 190.64.204.60:5560 > ;rport=5560;received=190.64.204.60;branch=z9hG4bK27ZB28p0y5mpr > Call-ID: 7c590ee4-30a4-1237-4894-00505601018a > From: "anacelia" ;tag=ejcX3jHgyFyrF > To: ;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R > CSeq: 128012877 BYE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1062 bytes from udp/[186.51.236.123]:38474 at 17:31:53.537993: > ------------------------------------------------------------------------ > REGISTER sip:190.64.204.60:5560 SIP/2.0 > Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.FMbxN6bPS;rport > From: ;tag=MBKS11QUc > To: sip:anacelia at 190.64.204.60 > CSeq: 24 REGISTER > Call-ID: DFvn85sO3E > Max-Forwards: 70 > Supported: replaces, outbound, gruu > Accept: application/sdp > Accept: text/plain > Accept: application/vnd.gsma.rcs-ft-http+xml > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="";+org.linphone.specs=groupchat > Expires: 3600 > User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) > Authorization: Digest realm="190.64.204.60", > nonce="6fce67e0-b601-11e8-982b-cdce49bc5d42", algorithm=MD5, > username="anacelia", uri="sip:190.64.204.60:5560", > response="077e0b6c433066db828f9ec829f8d558", cnonce="~Oh2sj~-KZHHDh9I", > nc=00000004, qop=auth > > ------------------------------------------------------------------------ > send 765 bytes to udp/[186.51.236.123]:38474 at 17:31:53.540658: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.45.210.97:39245 > ;branch=z9hG4bK.FMbxN6bPS;rport=38474;received=186.51.236.123 > From: ;tag=MBKS11QUc > To: ;tag=HDr7833tNa1gj > Call-ID: DFvn85sO3E > CSeq: 24 REGISTER > Contact: ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;expires=30 > Date: Tue, 11 Sep 2018 20:31:53 GMT > User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Sep 12 13:37:19 2018 From: brian at freeswitch.com (Brian West) Date: Wed, 12 Sep 2018 08:37:19 -0500 Subject: [Freeswitch-users] No ringtone to caller when third party call control (3PCC) is enabled In-Reply-To: <8d946ef6bbaa47319a2712a8270d3cee@enghouse.com> References: <8d946ef6bbaa47319a2712a8270d3cee@enghouse.com> Message-ID: Yes please submit a JIRA. On Wed, Sep 12, 2018 at 2:38 AM, Jaan Kaja wrote: > Hi, > > > > I have configured "enable-3pcc=proxy" in the SIP profile, and have an > external application that controls answer-call, make-call, etc. The > configuration means that incoming calls w/o SDP are accepted, which works > fine. > > The problem is that when I make an outgoing call, ringtone is not played > to the caller, although the “ringback” variable is set for the caller's leg. > > > > I have checked the code in mod_sofia, and in sofia.c, there is a piece of > code: > > if > (sofia_test_flag(tech_pvt, TFLAG_SKIP_EARLY)) { > > > sofia_clear_flag_locked(tech_pvt, TFLAG_SKIP_EARLY); > > goto > done; > > } > > which disables reaching the ring-ready state. TFLAG_SKIP_EARLY is set when > TFLAG_3PCC is true. If I comment out the goto statement, the A-party gets > ringtone. N.B.: This is just a minimal quick fix, not a proposed solution, > which works because I’m not doing an enterprise originate. > > > > Question: Is there a reason for this behavior? Documentation shows that > early media must be disabled when an enterprise originate is done, but why > is playing ringtone disabled in this case? Should I submit a bug report to > Jira? > > > > Best regards, > > Jaan > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.zenzerovic at inet.hr Wed Sep 12 12:17:28 2018 From: sebastian.zenzerovic at inet.hr (=?UTF-8?Q?Sebastian_Zenzerovi=c4=87?=) Date: Wed, 12 Sep 2018 14:17:28 +0200 Subject: [Freeswitch-users] Gateway SIP OPTIONS ping Contact header Message-ID: Hello, is it possible to add (configuration option/variable?) Contact header to outgoing gateway ping messages sent by Freeswitch? Remote SIP peer is answering "403 forbidden" as there is no Contact header present in SIP OPTIONS. Not a big deal since FS sees the peer as available/UP anyway but could be a interoperability enhancement between FS and Microsoft Direct routing service (https://aka.ms/dr/). Here are the requirements: • Must: When placing calls to the Direct Routing interface, the 'CONTACT' header must have the SBC FQDN in the URI hostname • Syntax: Contact: @:; • If the parameter is not configured correctly, OPTIONS are rejected with a '403 Forbidden' message Freeswitch log:    OPTIONS sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls SIP/2.0    Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS    Max-Forwards: 70    From: ;tag=0aB5jm3SF07ve    To:    Call-ID: 101b723c-3113-1237-6092-00155d000d46    CSeq: 128016309 OPTIONS    User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY    Supported: timer, path, replaces    Allow-Events: talk, hold, conference, refer    Content-Length: 0    SIP/2.0 403 Forbidden    FROM: ;tag=0aB5jm3SF07ve    TO:    CSEQ: 128016309 OPTIONS    CALL-ID: 101b723c-3113-1237-6092-00155d000d46    VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS    REASON: Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route and Contact headers are missing"    CONTENT-LENGTH: 0    ALLOW: INVITE    ALLOW: ACK    ALLOW: OPTIONS    ALLOW: CANCEL    ALLOW: BYE    ALLOW: NOTIFY    SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4 Thanks. BR, Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Sep 12 16:34:10 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Sep 2018 12:34:10 -0400 Subject: [Freeswitch-users] Gateway SIP OPTIONS ping Contact header In-Reply-To: References: Message-ID: It may be possible with code to just always add a contact, not sure there is any reason not to, just isn’t typically necessary so it wasn’t there. > On Sep 12, 2018, at 8:17 AM, Sebastian Zenzerović wrote: > > > Hello, > > is it possible to add (configuration option/variable?) Contact header to outgoing gateway ping messages sent by Freeswitch? > > Remote SIP peer is answering "403 forbidden" as there is no Contact header present in SIP OPTIONS. > Not a big deal since FS sees the peer as available/UP anyway but could be a interoperability enhancement between FS and Microsoft Direct routing service (https://aka.ms/dr/ ). > > Here are the requirements: > • Must: When placing calls to the Direct Routing interface, the 'CONTACT' header must have the SBC FQDN in the URI hostname > • Syntax: Contact: @:; > • If the parameter is not configured correctly, OPTIONS are rejected with a '403 Forbidden' message > > Freeswitch log: > > OPTIONS sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls SIP/2.0 > Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS > Max-Forwards: 70 > From: ;tag=0aB5jm3SF07ve > To: > Call-ID: 101b723c-3113-1237-6092-00155d000d46 > CSeq: 128016309 OPTIONS > User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Length: 0 > > SIP/2.0 403 Forbidden > FROM: ;tag=0aB5jm3SF07ve > TO: > CSEQ: 128016309 OPTIONS > CALL-ID: 101b723c-3113-1237-6092-00155d000d46 > VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS > REASON: Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route and Contact headers are missing" > CONTENT-LENGTH: 0 > ALLOW: INVITE > ALLOW: ACK > ALLOW: OPTIONS > ALLOW: CANCEL > ALLOW: BYE > ALLOW: NOTIFY > SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Sep 12 16:35:56 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Sep 2018 12:35:56 -0400 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI In-Reply-To: References: Message-ID: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> It’s failing due to no number to redirect to. This orgid thing must be some info about where it is supposed to be redirected to? This isn’t something that is currently supported, but that doesn’t mean there isn’t a way to add it. Do you have any info about how it should be routed? This would be something I’d be willing to entertain pull requests to add support for. Mike > On Sep 11, 2018, at 2:33 PM, Ben Kaufman wrote: > > Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user. > > The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event. > > In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000): > >> REFER-TO: > > In the case of transfer to another Teams user, the transfer Refer-To: header looks like this: > >> REFER-TO: > > When this occurs, the last line in the Freeswitch log for the call is: > >> [DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com] > > > There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report. > From ben.kaufman at altigen.com Wed Sep 12 19:08:32 2018 From: ben.kaufman at altigen.com (Ben Kaufman) Date: Wed, 12 Sep 2018 19:08:32 +0000 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI In-Reply-To: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> Message-ID: It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data: REFERRED-BY: I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan. Ben Kaufman ben.kaufman at altigen.com Cloud Operations Manager AltiGen Communications, Inc. -----Original Message----- From: FreeSWITCH-users On Behalf Of Michael Jerris Sent: Wednesday, September 12, 2018 11:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI It’s failing due to no number to redirect to. This orgid thing must be some info about where it is supposed to be redirected to? This isn’t something that is currently supported, but that doesn’t mean there isn’t a way to add it. Do you have any info about how it should be routed? This would be something I’d be willing to entertain pull requests to add support for. Mike > On Sep 11, 2018, at 2:33 PM, Ben Kaufman wrote: > > Setting up Freeswitch to connect to Microsoft Teams' "Direct Routing" (https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan). General call setup-teardown is successful, but I'm encountering a problem with call transfers if the transfer target is another Microsoft Teams user. > > The most immediate obvious difference between the transfer to an external party and the transfer to another Teams user is in the Refer-To header of the Refer message on the transfer event. > > In the case of transfer to an external party, the Refer-To is (phone number redacted to 555-555-1000): > >> REFER-TO: > > In the case of transfer to another Teams user, the transfer Refer-To: header looks like this: > >> REFER-TO: > > When this occurs, the last line in the Freeswitch log for the call is: > >> [DEBUG] sofia.c:8544 Process REFER to [(null)@sip.pstnhub.microsoft.com] > > > There is nothing else in the log until this times out, no indication it has entered any dialplan, etc. Even on timeout, it is just the teams client saying that the call has failed, and unless one end hangs up, or reconnects to the failed transfer, there is no action in freeswitch. I believe this to be a bug, but wanted to get an opinion before opening a formal bug report. > _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator at altigen.com, and destroy all copies of this message and any attachments. From mike at jerris.com Wed Sep 12 19:15:30 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Sep 2018 15:15:30 -0400 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI In-Reply-To: References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> Message-ID: <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> I’m open to patches that add this functionality. Mike > On Sep 12, 2018, at 3:08 PM, Ben Kaufman wrote: > > It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data: > > REFERRED-BY: > > > I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan. > From brian at freeswitch.com Wed Sep 12 22:15:06 2018 From: brian at freeswitch.com (Brian West) Date: Wed, 12 Sep 2018 17:15:06 -0500 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI In-Reply-To: <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> Message-ID: Ben, Thanks for talking to me about this today, can you file a jira with the details of this at freeswitch.org/jira? Thanks, /b On Wed, Sep 12, 2018 at 2:15 PM, Michael Jerris wrote: > I’m open to patches that add this functionality. > > Mike > > > On Sep 12, 2018, at 3:08 PM, Ben Kaufman > wrote: > > > > It's possibly the orgid. There's also a Referred-By: header that > contains a large amount of encoded (or encrypted? It's not base 64) data: > > > > REFERRED-BY: 1c14-454b-b910-2e20b20b2ab7;x-t=a3217d98-07a6-45fc-8c9c-ce427f5ed8ac;x-a= > qfNyIy1nfSvIYN63RNEJ1hXKpiLor84ceFqs6iAWusN3wcw5S5tDDiugmb% > 2b9hOLxcObLtp6oAUPkCDqTCPAGfQtCqvbDlTpTSG7YZHZCSuhvMUB% > 2fwYVDmbkR1COvm72MtszpjCwwBwkNFxRtJB0hsdiprRrQbvZylULQpIcLQK > CxPO8U6Rah7XKM1B8hnJGC66ok3gh9yy0J0q3vCKVwC6w7yhz5PxUVswSkhj > qvnXq1MRlF4hTfCIP%2f%2bz34EOQwEowWwXJftnYQkSd1K3tjT5i9VdmiZLg%2bgP0znqk% > 2fQadRE6QdwuABe9oSALATtl808KuviwbZG7fftIeUi4xsrVkO% > 2bNfwUseeK9YZ8mxUFd2oStRaGVo4ualIZztAti5rEc179dPnw6vpdZbyZMB > GmYOjYB%2bN4AbzQsgXGiTMl%2b5Ujqjozeb0L2%2bx7VAoDdkfa%2buGhkFlLi% > 2buu3r0gEZo7vdxa44FJYyZFrRRGES1PFgkIB0egbG6HBewkI0ucggNKBlzC > 4r4fjD8IZVE%2bmovyD95sAehns%2bEuhXsBFek8JXI2vnVlYIYyCLNdcFTq9jyUomMze7% > 2bNK86UX6W6a75x20srK1f0qERu0VN2%2bVVV4P2CQuXWiyIve% > 2b82CEzN9T4oO8m0ZgIB3s%2fIK%2boZRDTVWZD1FJ6AdBIS8S7qi1kTmV > Zds45Sh97vzKHnhETYxe3XMRn78SOP9oYJgyfr814CfTC0EwQzZ8jdjVa0Hs > XIDu6OsT7fOAwtbdTDovntt9J> > > > > > > I have a Sonus box that does handle the REFER in a manner where the > INVITE to establish with this header, as well as the aforementioned > x-m-8:orgid:.... parameter in the rURI, but I think carrying things over > like that could be done in the dialplan (I think). At this point, it > doesn't enter the dialplan. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.zenzerovic at inet.hr Thu Sep 13 07:46:46 2018 From: sebastian.zenzerovic at inet.hr (=?UTF-8?Q?Sebastian_Zenzerovi=c4=87?=) Date: Thu, 13 Sep 2018 09:46:46 +0200 Subject: [Freeswitch-users] Gateway SIP OPTIONS ping Contact header In-Reply-To: References: Message-ID: <8644c096-708f-0b8d-7001-478f503ce2b2@inet.hr> Hello, should I open a ticket in Jira (issue type: improvement?) if I'd like to have this implemented in future versions? BR, Sebastian On 12.9.2018. 18:34, Michael Jerris wrote: > It may be possible with code to just always add a contact, not sure > there is any reason not to, just isn’t typically necessary so it > wasn’t there. > >> On Sep 12, 2018, at 8:17 AM, Sebastian Zenzerović >> > >> wrote: >> >> >> Hello, >> >> is it possible to add (configuration option/variable?) Contact header >> to outgoing gateway ping messages sent by Freeswitch? >> >> Remote SIP peer is answering "403 forbidden" as there is no Contact >> header present in SIP OPTIONS. >> Not a big deal since FS sees the peer as available/UP anyway but >> could be a interoperability enhancement between FS and Microsoft >> Direct routing service (https://aka.ms/dr/). >> >> Here are the requirements: >> • Must: When placing calls to the Direct Routing interface, the >> 'CONTACT' header must have the SBC FQDN in the URI hostname >> • Syntax: Contact: @:> Port>; >> • If the parameter is not configured correctly, OPTIONS are rejected >> with a '403 Forbidden' message >> >> Freeswitch log: >> >>    OPTIONS sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls SIP/2.0 >>    Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS >>    Max-Forwards: 70 >>    From: ;tag=0aB5jm3SF07ve >>    To: >>    Call-ID: 101b723c-3113-1237-6092-00155d000d46 >>    CSeq: 128016309 OPTIONS >>    User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit >>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >>    Supported: timer, path, replaces >>    Allow-Events: talk, hold, conference, refer >>    Content-Length: 0 >> >>    SIP/2.0 403 Forbidden >>    FROM: ;tag=0aB5jm3SF07ve >>    TO: >>    CSEQ: 128016309 OPTIONS >>    CALL-ID: 101b723c-3113-1237-6092-00155d000d46 >>    VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS >>    REASON: >> Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route >> and Contact headers are missing" >>    CONTENT-LENGTH: 0 >>    ALLOW: INVITE >>    ALLOW: ACK >>    ALLOW: OPTIONS >>    ALLOW: CANCEL >>    ALLOW: BYE >>    ALLOW: NOTIFY >>    SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4 > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Sep 13 08:03:20 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 13 Sep 2018 08:03:20 +0000 Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? In-Reply-To: <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> , <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> Message-ID: Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Thu Sep 13 13:07:03 2018 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 13 Sep 2018 14:07:03 +0100 Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? In-Reply-To: References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> Message-ID: Try setting rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF and enable the jitterbuffer by setting the channel variable jitterbuffer_msec=60. This is the solution to https://freeswitch.org/jira/browse/FS-4905 On Thu, 13 Sep 2018 at 13:26, Shaun Stokes wrote: > Hi All, > > > Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP > stream on an inbound call from a carrier so that this isn't audible in the > call recording while still retaining DTMF Outofband (RFC2833) in the SIP > messages? If not, how could this be achieved? > > > Thanks, > > Shaun > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Thu Sep 13 16:10:28 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 13 Sep 2018 16:10:28 +0000 (UTC) Subject: How mod_verto is invoked to handle incoming call? References: <814451549.4786384.1536855028245.ref@mail.yahoo.com> Message-ID: <814451549.4786384.1536855028245@mail.yahoo.com> Hi all, I am learning Freeswitch webrtc, and setup an environment with demo configuration as below. 1. User 1008 is logged in from browser with Verto. 2. User 1000 is logged in from X-Lite SIP phone. "sofia status profile internal reg" only shows 1000. 1000 makes a call to 1008, 1008 answers the call, and call is established between browser and X-Lite. The following is from freeswitch log, 2662 7fbee0e2-b766-11e8-b3d6-f9bb4821b253 2018-09-13 15:05:49.296720 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2663 7fc69cd8-b766-11e8-b3f8-f9bb4821b253 2018-09-13 15:05:49.296720 [DEBUG] mod_rtc.c:392 () State Change CS_NEW -> CS_INIT I am wondering how mod_verto is invoked to handle the incoming call. Can someone explain please? Many thanks, /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From ryandelgrosso at gmail.com Thu Sep 13 18:36:31 2018 From: ryandelgrosso at gmail.com (Ryan Delgrosso) Date: Thu, 13 Sep 2018 11:36:31 -0700 Subject: [Freeswitch-users] Date Time parsing question Message-ID: Hello all, Ive been puzzling over a date time parsing issue in FS and maybe someone can tell me what ive done wrong, since this log message makes little sense: XML DateTime Check: date time[2018-09-13 12:27:50] =~ 2018-09-13 13:00~2018-09-18 08:00 (PASS) I would expect that to evaluate to false/fail since the datetime is not between the specified parameters. FWIW I also tried specifying seconds in the argument with the same result. Any hints are much appreciated. -Ryan From shaun.stokes at itec-support.co.uk Fri Sep 14 07:23:40 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 14 Sep 2018 07:23:40 +0000 Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? In-Reply-To: References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> , <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com>, Message-ID: After some research we've found that this may achieve the desired results: We will post back with the results after testing. Sources: http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/121194.html http://lists.freeswitch.org/pipermail/freeswitch-users/2016-July/121356.html ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 13 September 2018 09:03:20 To: FreeSWITCH Users Help Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From ravindra.bhatt at ecosmob.com Mon Sep 17 05:55:05 2018 From: ravindra.bhatt at ecosmob.com (Ravindrakumar Bhatt) Date: Mon, 17 Sep 2018 11:25:05 +0530 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call Message-ID: Hello, I am using freeswitch 1.6.20. it is fresh setup with basic configuration. i want to renegotiate codec when call is answered and i am using uuid_media_reneg for same. SCENARIO: user a calls to user b and both user are using code G722. when i use uuid_meida_reneg to change codec to G729 for A leg it is working fine but when using same api with bleg uuid for B leg it is using G722 for renegotiate. Here are console log for same. https://pastebin.freeswitch.org/view/342e4eed Any suggestions for this or setting needed to be done in freeswitch. -- *Thanks and Regards,* *Ravindrakumar Bhatt* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+918460692402* -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Mon Sep 17 19:33:46 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 17 Sep 2018 12:33:46 -0700 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Hi Ravindrakumar, Why don't you limit them to G729 to begin with instead of changing them once the call is established? On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < ravindra.bhatt at ecosmob.com> wrote: > > Hello, > I am using freeswitch 1.6.20. it is fresh setup with basic configuration. > > i want to renegotiate codec when call is answered and i am using > uuid_media_reneg for same. > > SCENARIO: > user a calls to user b and both user are using code G722. when i use > uuid_meida_reneg to change codec to G729 for A leg it is working fine but > when using same api with bleg uuid for B leg it is using G722 for > renegotiate. > Here are console log for same. > https://pastebin.freeswitch.org/view/342e4eed > Any suggestions for this or setting needed to be done in freeswitch. > > -- > > *Thanks and Regards,* > > *Ravindrakumar Bhatt* > Jr. Software Developer > Ecosmob Technologies Ltd > Ahmedabad > Mo:*+918460692402* > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Tue Sep 18 08:36:10 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 18 Sep 2018 08:36:10 +0000 Subject: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? In-Reply-To: References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> , Message-ID: Thanks Steve, we'll give this a try. ________________________________ From: FreeSWITCH-users on behalf of Steven Ayre Sent: 13 September 2018 14:07:03 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to drop Inband DTMF tones while retaining Outofband DTMF tones? Try setting rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF and enable the jitterbuffer by setting the channel variable jitterbuffer_msec=60. This is the solution to https://freeswitch.org/jira/browse/FS-4905 On Thu, 13 Sep 2018 at 13:26, Shaun Stokes > wrote: Hi All, Is it possible for FreeSWITCH to drop Inband DTMF tones from the RTP\SRTP stream on an inbound call from a carrier so that this isn't audible in the call recording while still retaining DTMF Outofband (RFC2833) in the SIP messages? If not, how could this be achieved? Thanks, Shaun _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bhattravi4 at gmail.com Tue Sep 18 09:36:24 2018 From: bhattravi4 at gmail.com (RAVI bhatt) Date: Tue, 18 Sep 2018 15:06:24 +0530 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Hi Joel, I am providing only g722 and g729 both codec to user. if both users uses same codec it's ok but if either of user is using G729 than i want to renegotiate g729 as audio codec for both leg instead of using transcoding.Here G722 is high bandwidth and g729 is low bandwidth. On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: > Hi Ravindrakumar, > > Why don't you limit them to G729 to begin with instead of changing them > once the call is established? > > On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < > ravindra.bhatt at ecosmob.com> wrote: > >> >> Hello, >> I am using freeswitch 1.6.20. it is fresh setup with basic configuration. >> >> i want to renegotiate codec when call is answered and i am using >> uuid_media_reneg for same. >> >> SCENARIO: >> user a calls to user b and both user are using code G722. when i use >> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >> when using same api with bleg uuid for B leg it is using G722 for >> renegotiate. >> Here are console log for same. >> https://pastebin.freeswitch.org/view/342e4eed >> Any suggestions for this or setting needed to be done in freeswitch. >> >> -- >> >> *Thanks and Regards,* >> >> *Ravindrakumar Bhatt* >> Jr. Software Developer >> Ecosmob Technologies Ltd >> Ahmedabad >> Mo:*+918460692402* >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Sep 19 17:16:57 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 19 Sep 2018 13:16:57 -0400 Subject: [Freeswitch-users] How mod_verto is invoked to handle incoming call? In-Reply-To: References: <814451549.4786384.1536855028245.ref@mail.yahoo.com> Message-ID: I’ve answered your question on this week’s community corner on cluecon weekly. > On Sep 14, 2018, at 11:07 AM, kaiduan xie via FreeSWITCH-users wrote: > > > From: kaiduan xie > Subject: How mod_verto is invoked to handle incoming call? > Date: September 13, 2018 at 12:10:28 PM EDT > To: FreeSWITCH Users Help > > > Hi all, > > I am learning Freeswitch webrtc, and setup an environment with demo configuration as below. > > 1. User 1008 is logged in from browser with Verto. > > 2. User 1000 is logged in from X-Lite SIP phone. > > "sofia status profile internal reg" only shows 1000. > > 1000 makes a call to 1008, 1008 answers the call, and call is established between browser and X-Lite. > > The following is from freeswitch log, > 2662 7fbee0e2-b766-11e8-b3d6-f9bb4821b253 2018-09-13 15:05:49.296720 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > > 2663 7fc69cd8-b766-11e8-b3f8-f9bb4821b253 2018-09-13 15:05:49.296720 [DEBUG] mod_rtc.c:392 () State Change CS_NEW -> CS_INIT > > I am wondering how mod_verto is invoked to handle the incoming call. Can someone explain please? > > Many thanks, > > /Kaiduan > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Tue Sep 18 20:46:59 2018 From: joel at textplus.com (Joel Serrano) Date: Tue, 18 Sep 2018 13:46:59 -0700 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Have a look at inbound-late-negotiation: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/2883752 I think it might help you On Tue, Sep 18, 2018 at 13:21 RAVI bhatt wrote: > Hi Joel, > I am providing only g722 and g729 both codec to user. if both users uses > same codec it's ok but if either of user is using G729 than i want to > renegotiate g729 as audio codec for both leg instead of using > transcoding.Here G722 is high bandwidth and g729 is low bandwidth. > > On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: > >> Hi Ravindrakumar, >> >> Why don't you limit them to G729 to begin with instead of changing them >> once the call is established? >> >> On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < >> ravindra.bhatt at ecosmob.com> wrote: >> >>> >>> Hello, >>> I am using freeswitch 1.6.20. it is fresh setup with basic configuration. >>> >>> i want to renegotiate codec when call is answered and i am using >>> uuid_media_reneg for same. >>> >>> SCENARIO: >>> user a calls to user b and both user are using code G722. when i use >>> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >>> when using same api with bleg uuid for B leg it is using G722 for >>> renegotiate. >>> Here are console log for same. >>> https://pastebin.freeswitch.org/view/342e4eed >>> Any suggestions for this or setting needed to be done in freeswitch. >>> >>> -- >>> >>> *Thanks and Regards,* >>> >>> *Ravindrakumar Bhatt* >>> Jr. Software Developer >>> Ecosmob Technologies Ltd >>> Ahmedabad >>> Mo:*+918460692402* >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Tue Sep 18 22:28:52 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 18 Sep 2018 19:28:52 -0300 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: If you offer the codecs in your preferred order (G.729, G.722), and both clients support them, they should choose G729. from RFC 3264: In all cases, the formats in the "m=" line MUST be listed in order of preference, with the first format listed being preferred. In this case, preferred means that the recipient of the offer SHOULD use the format with the highest preference that is acceptable to it. Guillermo On Tue, Sep 18, 2018 at 6:06 PM RAVI bhatt wrote: > Hi Joel, > I am providing only g722 and g729 both codec to user. if both users uses > same codec it's ok but if either of user is using G729 than i want to > renegotiate g729 as audio codec for both leg instead of using > transcoding.Here G722 is high bandwidth and g729 is low bandwidth. > > On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: > >> Hi Ravindrakumar, >> >> Why don't you limit them to G729 to begin with instead of changing them >> once the call is established? >> >> On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < >> ravindra.bhatt at ecosmob.com> wrote: >> >>> >>> Hello, >>> I am using freeswitch 1.6.20. it is fresh setup with basic configuration. >>> >>> i want to renegotiate codec when call is answered and i am using >>> uuid_media_reneg for same. >>> >>> SCENARIO: >>> user a calls to user b and both user are using code G722. when i use >>> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >>> when using same api with bleg uuid for B leg it is using G722 for >>> renegotiate. >>> Here are console log for same. >>> https://pastebin.freeswitch.org/view/342e4eed >>> Any suggestions for this or setting needed to be done in freeswitch. >>> >>> -- >>> >>> *Thanks and Regards,* >>> >>> *Ravindrakumar Bhatt* >>> Jr. Software Developer >>> Ecosmob Technologies Ltd >>> Ahmedabad >>> Mo:*+918460692402* >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 19 07:15:16 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 19 Sep 2018 08:15:16 +0100 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Isn't that what priorities are for? You offer first 729, then 722. If both ends have both, they should use 729. On Tue, Sep 18, 2018, 22:13 RAVI bhatt wrote: > Hi Joel, > I am providing only g722 and g729 both codec to user. if both users uses > same codec it's ok but if either of user is using G729 than i want to > renegotiate g729 as audio codec for both leg instead of using > transcoding.Here G722 is high bandwidth and g729 is low bandwidth. > > On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: > >> Hi Ravindrakumar, >> >> Why don't you limit them to G729 to begin with instead of changing them >> once the call is established? >> >> On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < >> ravindra.bhatt at ecosmob.com> wrote: >> >>> >>> Hello, >>> I am using freeswitch 1.6.20. it is fresh setup with basic configuration. >>> >>> i want to renegotiate codec when call is answered and i am using >>> uuid_media_reneg for same. >>> >>> SCENARIO: >>> user a calls to user b and both user are using code G722. when i use >>> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >>> when using same api with bleg uuid for B leg it is using G722 for >>> renegotiate. >>> Here are console log for same. >>> https://pastebin.freeswitch.org/view/342e4eed >>> Any suggestions for this or setting needed to be done in freeswitch. >>> >>> -- >>> >>> *Thanks and Regards,* >>> >>> *Ravindrakumar Bhatt* >>> Jr. Software Developer >>> Ecosmob Technologies Ltd >>> Ahmedabad >>> Mo:*+918460692402* >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed Sep 19 16:31:26 2018 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 19 Sep 2018 20:31:26 +0400 Subject: [Freeswitch-users] ring all devices other than self with registration id as same Message-ID: hi, i have a client registered to FS using same id on 2 devices and i created a dialplan entry such that if he calls 000 then it can ring all devices registered with same id, problem is the call also comes to himself on the device he calls 000 from, is there a way to ring all other devices registered other than his (multiple registration is enabled on profile) the bridge statement im using is as below currently where 5656 is his registered userid -- Regards, Bipin -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Wed Sep 19 17:46:55 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Wed, 19 Sep 2018 17:46:55 +0000 (UTC) Subject: [Freeswitch-users] How mod_verto is invoked to handle incoming call? In-Reply-To: References: <814451549.4786384.1536855028245.ref@mail.yahoo.com> Message-ID: <1476917116.8284303.1537379215912@mail.yahoo.com> Michael, Where can find the answer in community corner on cluecon weekly? Thanks, /Kaiduan On Wednesday, September 19, 2018, 1:17:01 p.m. EDT, Michael Jerris wrote: I’ve answered your question on this week’s community corner on cluecon weekly. On Sep 14, 2018, at 11:07 AM, kaiduan xie via FreeSWITCH-users wrote: From: kaiduan xie Subject: How mod_verto is invoked to handle incoming call? Date: September 13, 2018 at 12:10:28 PM EDT To: FreeSWITCH Users Help Hi all, I am learning Freeswitch webrtc, and setup an environment with demo configuration as below. 1. User 1008 is logged in from browser with Verto. 2. User 1000 is logged in from X-Lite SIP phone. "sofia status profile internal reg" only shows 1000. 1000 makes a call to 1008, 1008 answers the call, and call is established between browser and X-Lite. The following is from freeswitch log, 2662 7fbee0e2-b766-11e8-b3d6-f9bb4821b253 2018-09-13 15:05:49.296720 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2663 7fc69cd8-b766-11e8-b3f8-f9bb4821b253 2018-09-13 15:05:49.296720 [DEBUG] mod_rtc.c:392 () State Change CS_NEW -> CS_INIT I am wondering how mod_verto is invoked to handle the incoming call. Can someone explain please? Many thanks, /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From bhattravi4 at gmail.com Thu Sep 20 02:44:31 2018 From: bhattravi4 at gmail.com (RAVI bhatt) Date: Thu, 20 Sep 2018 08:14:31 +0530 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Thanks to all for your replies. I am using priority for supporting codec and it's working fine for most of scenarios but it is not working for one scenario where aleg has priority as g722,g729 while bleg has g729,g722.In this scenario g722 is negotiated but as low bandwidth codec I want to provide G729 as negotiated codec. And this is the main question how should I achieve this? because I am not able to renegotiate codecs using uuid_media_reneg. On Thu 20 Sep, 2018, 12:59 AM David Villasmil, < david.villasmil.work at gmail.com> wrote: > Isn't that what priorities are for? > You offer first 729, then 722. If both ends have both, they should use 729. > > On Tue, Sep 18, 2018, 22:13 RAVI bhatt wrote: > >> Hi Joel, >> I am providing only g722 and g729 both codec to user. if both users uses >> same codec it's ok but if either of user is using G729 than i want to >> renegotiate g729 as audio codec for both leg instead of using >> transcoding.Here G722 is high bandwidth and g729 is low bandwidth. >> >> On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: >> >>> Hi Ravindrakumar, >>> >>> Why don't you limit them to G729 to begin with instead of changing them >>> once the call is established? >>> >>> On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < >>> ravindra.bhatt at ecosmob.com> wrote: >>> >>>> >>>> Hello, >>>> I am using freeswitch 1.6.20. it is fresh setup with basic >>>> configuration. >>>> >>>> i want to renegotiate codec when call is answered and i am using >>>> uuid_media_reneg for same. >>>> >>>> SCENARIO: >>>> user a calls to user b and both user are using code G722. when i use >>>> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >>>> when using same api with bleg uuid for B leg it is using G722 for >>>> renegotiate. >>>> Here are console log for same. >>>> https://pastebin.freeswitch.org/view/342e4eed >>>> Any suggestions for this or setting needed to be done in freeswitch. >>>> >>>> -- >>>> >>>> *Thanks and Regards,* >>>> >>>> *Ravindrakumar Bhatt* >>>> Jr. Software Developer >>>> Ecosmob Technologies Ltd >>>> Ahmedabad >>>> Mo:*+918460692402* >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From francisco.caraballo at cyberfonica.com Thu Sep 20 10:02:38 2018 From: francisco.caraballo at cyberfonica.com (Francisco Caraballo) Date: Thu, 20 Sep 2018 12:02:38 +0200 Subject: [Freeswitch-users] http_cache use of local cache directory Message-ID: Hello. I'm trying to figure out how mod_htp_cache uses the local cache directory when uploading a file to S3. Speciffically I would like to know how long will the file be kept in the local file system cache (by default /usr/local/freeswitch/cache) in the case of an error while uploading to S3. Will it be deleted next time freeswitch starts? Will it stay there forever? The reason I'm asking is because we are trying to implement a fallback system to recover and upload recordings to S3 in case there was a problem when http_cache first attempted to upload the file (for example a 403 error, or a connectivity issue). Thanks a lot for your help. Francisco Caraballo -------------- next part -------------- An HTML attachment was scrubbed... URL: From ben.kaufman at altigen.com Thu Sep 20 14:54:22 2018 From: ben.kaufman at altigen.com (Ben Kaufman) Date: Thu, 20 Sep 2018 14:54:22 +0000 Subject: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI In-Reply-To: References: <43FAFA93-D2A2-431A-B523-562605FEF97A@jerris.com> <1DCDB595-93E7-476E-9C01-98AE5F30FE60@jerris.com> Message-ID: I had a lot of trouble with the act of opening a ticket. It’s open now (FS-11401), however any attempt to attach my log files failed. Additionally, it looks like I’m not able to post an update to the ticket either to explain this. I have logs for this, but when i attempt to upload them, Jira gives an error that the upload was not successful and that "An internal error has occured. Please contact your administrator". The log files are less than 200 KB in size,so it's likely not a size issue. I also have a log from a Sonus SBC that does process the transfer which could be used as a comparison of the full dialog. Even when I log into JIRA, it even gives an error: We've detected a potential problem with JIRA's Dashboard configuration that your administrator can correct. Hide Dashboard Diagnostics: Mismatched URL Hostname JIRA is reporting that it is running on the hostname 'freeswitch.org', which does not match the hostname used to run these diagnostics, 'freeswitch.com'. This is known to cause JIRA to construct URLs using the incorrect hostname, which will result in errors in the dashboard, among other issues. The most common cause of this is the use of a reverse-proxy HTTP server (often Apache or IIS) in front of the application server running JIRA. While this configuration is supported, some additional setup might be necessary in order to ensure that JIRA detects the correct hostname. The following articles describe the issue and the steps you should take to ensure that your web server and app server are configured correctly: * Gadgets do not display correctly after upgrade to JIRA 4.0 * Integrating JIRA with Apache * Integrating JIRA with Apache using SSL If you believe this diagnosis is in error, or you have any other questions, please contact Atlassian Support. Detailed Error Click here to learn more Ben Kaufman ben.kaufman at altigen.com Cloud Operations Manager AltiGen Communications, Inc. From: FreeSWITCH-users On Behalf Of Brian West Sent: Wednesday, September 12, 2018 5:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Transfer fails (does nothing) when receiving REFER message with no user in the Refer-To: URI Ben, Thanks for talking to me about this today, can you file a jira with the details of this at freeswitch.org/jira? Thanks, /b On Wed, Sep 12, 2018 at 2:15 PM, Michael Jerris > wrote: I’m open to patches that add this functionality. Mike > On Sep 12, 2018, at 3:08 PM, Ben Kaufman > wrote: > > It's possibly the orgid. There's also a Referred-By: header that contains a large amount of encoded (or encrypted? It's not base 64) data: > > REFERRED-BY: > > > I have a Sonus box that does handle the REFER in a manner where the INVITE to establish with this header, as well as the aforementioned x-m-8:orgid:.... parameter in the rURI, but I think carrying things over like that could be done in the dialplan (I think). At this point, it doesn't enter the dialplan. > _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- [Image removed by sender.] Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [Image removed by sender. https://www.facebook.com/signalwireinc?src=email][Image removed by sender. https://twitter.com/freeswitch] STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator at altigen.com, and destroy all copies of this message and any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 596 bytes Desc: image001.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 350 bytes Desc: image002.jpg URL: From brian at freeswitch.com Thu Sep 20 15:46:56 2018 From: brian at freeswitch.com (Brian West) Date: Thu, 20 Sep 2018 10:46:56 -0500 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: No, this is not accurate, you can choose a lower codec in your answer if you choose to, this is why our codec negotiation prefs have a 'greedy' option. On Wed, Sep 19, 2018 at 2:15 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Isn't that what priorities are for? > You offer first 729, then 722. If both ends have both, they should use 729. > > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Thu Sep 20 19:38:14 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 20 Sep 2018 16:38:14 -0300 Subject: [Freeswitch-users] Facing issue while using uuid_media_reneg on answered call In-Reply-To: References: Message-ID: Did you try with and ? Guillermo On Thu, Sep 20, 2018 at 1:14 PM RAVI bhatt wrote: > Thanks to all for your replies. > > I am using priority for supporting codec and it's working fine for most of > scenarios but it is not working for one scenario where aleg has priority as > g722,g729 while bleg has g729,g722.In this scenario g722 is negotiated but > as low bandwidth codec I want to provide > G729 as negotiated codec. > > And this is the main question how should I achieve this? because I am not > able to renegotiate codecs using uuid_media_reneg. > > On Thu 20 Sep, 2018, 12:59 AM David Villasmil, < > david.villasmil.work at gmail.com> wrote: > >> Isn't that what priorities are for? >> You offer first 729, then 722. If both ends have both, they should use >> 729. >> >> On Tue, Sep 18, 2018, 22:13 RAVI bhatt wrote: >> >>> Hi Joel, >>> I am providing only g722 and g729 both codec to user. if both users uses >>> same codec it's ok but if either of user is using G729 than i want to >>> renegotiate g729 as audio codec for both leg instead of using >>> transcoding.Here G722 is high bandwidth and g729 is low bandwidth. >>> >>> On Tue 18 Sep, 2018, 2:29 AM Joel Serrano, wrote: >>> >>>> Hi Ravindrakumar, >>>> >>>> Why don't you limit them to G729 to begin with instead of changing them >>>> once the call is established? >>>> >>>> On Mon, Sep 17, 2018 at 7:46 AM Ravindrakumar Bhatt < >>>> ravindra.bhatt at ecosmob.com> wrote: >>>> >>>>> >>>>> Hello, >>>>> I am using freeswitch 1.6.20. it is fresh setup with basic >>>>> configuration. >>>>> >>>>> i want to renegotiate codec when call is answered and i am using >>>>> uuid_media_reneg for same. >>>>> >>>>> SCENARIO: >>>>> user a calls to user b and both user are using code G722. when i use >>>>> uuid_meida_reneg to change codec to G729 for A leg it is working fine but >>>>> when using same api with bleg uuid for B leg it is using G722 for >>>>> renegotiate. >>>>> Here are console log for same. >>>>> https://pastebin.freeswitch.org/view/342e4eed >>>>> Any suggestions for this or setting needed to be done in freeswitch. >>>>> >>>>> -- >>>>> >>>>> *Thanks and Regards,* >>>>> >>>>> *Ravindrakumar Bhatt* >>>>> Jr. Software Developer >>>>> Ecosmob Technologies Ltd >>>>> Ahmedabad >>>>> Mo:*+918460692402* >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Services >>>>> sales at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From splva at mail.ru Thu Sep 20 20:12:49 2018 From: splva at mail.ru (Ilya Spirin) Date: Thu, 20 Sep 2018 13:12:49 -0700 (MST) Subject: Detecting speech over ESL using mod_unimrcp. Message-ID: <1537474369458-0.post@n2.nabble.com> Hello everyone! I'm trying to recognize speech with my application. FS built with mod_unimrcp. Also I have installed UniMRCP Server with Google Speech Recognition plugin. play_and_detect_speech application works fine. But I have to stop playback manually that's why I'm sending some commands via ESL. Here is ESL dump: https://pastebin.com/p1jUrhn6 After sending answer, playback and detect_speech commands I immediately got "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. Please help. What I'm doing wrong? Thank you. -- Best regards Ilya Spirin -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From anthony.minessale at gmail.com Fri Sep 21 00:29:06 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 20 Sep 2018 19:29:06 -0500 Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. In-Reply-To: References: Message-ID: Its async so send the call to do something like park so it doesn't end. On Thu, Sep 20, 2018 at 4:21 PM Ilya Spirin via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Ilya Spirin > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 20 Sep 2018 13:12:49 -0700 (MST) > Subject: Detecting speech over ESL using mod_unimrcp. > Hello everyone! > > I'm trying to recognize speech with my application. FS built with > mod_unimrcp. Also I have installed UniMRCP Server with Google Speech > Recognition plugin. > > play_and_detect_speech application works fine. But I have to stop playback > manually that's why I'm sending some commands via ESL. > > Here is ESL dump: > https://pastebin.com/p1jUrhn6 > After sending answer, playback and detect_speech commands I immediately got > "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. > > Please help. What I'm doing wrong? > Thank you. > > -- > Best regards > Ilya Spirin > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: Ilya Spirin via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 20 Sep 2018 14:21:25 -0700 (PDT) > Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at smallunix.net Fri Sep 21 10:47:02 2018 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 21 Sep 2018 12:47:02 +0200 Subject: [Freeswitch-users] zRTP + OPUS one-way audio issue Message-ID: Hi, I'm facing an one-way audio issue with zRTP while using OPUS. zRTP is set in passthru mode. Called party can hear Calling party but not vice versa. Full log: https://pastebin.freeswitch.org/view/639adfe9 This issue is happening *only* with dynamic payloads codecs. Everything is working fine if I use G722 or G729. Scenario Leg-A (207) -> FS -> Leg-B (213) Checking FS logs, I see Leg-A's SDP: a=rtpmap:103 opus/48000/2 Leg-B's SDP: a=rtpmap:102 opus/48000/2 Actually having different payloads should not be an issue. I opened a case to Acrobits, they said the issue is in the actual RTP traffic sent to Leg-A from FS: >From the client log Leg-A Sending RTP packet #200 192.168.128.158:59176 > 31.102.111.134:28560, len=116, really=116, data=80679E55DF67E34A291167D899830CC03C124F353DA3C069574410EC0333125F This is using 103, correct as agreed on this side of the call Received RTP packet #200 31.102.111.134:28560 > 192.168.128.158:59176, len=90, data=80663D3B22CC8CB414490864A230F49B6EFF42E7BCA9EB64B864374B1377965C This is using 102, which is wrong for this side of the call. Seems that after negotiated payload 102 with Leg-B, FS is trying to use it on Leg-A, where it should be used 102 instead. Any idea? FreeSWITCH Version 1.8.1-2-4f54cff~64bit (-2-4f54cff 64bit) OS.: Debian 8.11 Linux pbx-186 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64 GNU/Linux Thank you, Andrea Mazzeo -------------- next part -------------- An HTML attachment was scrubbed... URL: From splva at mail.ru Fri Sep 21 13:16:56 2018 From: splva at mail.ru (Ilya Spirin) Date: Fri, 21 Sep 2018 16:16:56 +0300 Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. In-Reply-To: References: Message-ID: <767b898f-532f-4618-c49e-6a97777fe413@mail.ru> Thanks for reply. Could you tell more please? I send a playback command already and file is playing for some time. But exactly detect_speech application ends quickly. Tried to send "detect_speech resume". There is still no effect (see pastebin). On 21.09.2018 03:29, Anthony Minessale wrote: > Its async so send the call to do something like park so it doesn't end. > > On Thu, Sep 20, 2018 at 4:21 PM Ilya Spirin via FreeSWITCH-users > > wrote: > > > > > ---------- Forwarded message ---------- > From: Ilya Spirin > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Thu, 20 Sep 2018 13:12:49 -0700 (MST) > Subject: Detecting speech over ESL using mod_unimrcp. > Hello everyone! > > I'm trying to recognize speech with my application. FS built with > mod_unimrcp. Also I have installed UniMRCP Server with Google Speech > Recognition plugin. > > play_and_detect_speech application works fine. But I have to stop > playback > manually that's why I'm sending some commands via ESL. > > Here is ESL dump: > https://pastebin.com/p1jUrhn6 > After sending answer, playback and detect_speech commands I > immediately got > "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. > > Please help. What I'm doing wrong? > Thank you. > > -- > Best regards > Ilya Spirin > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: Ilya Spirin via FreeSWITCH-users > > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Thu, 20 Sep 2018 14:21:25 -0700 (PDT) > Subject: [Freeswitch-users] Detecting speech over ESL using > mod_unimrcp. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From nick.woodward at hotmail.co.uk Fri Sep 21 14:51:29 2018 From: nick.woodward at hotmail.co.uk (Nick Woodward) Date: Fri, 21 Sep 2018 14:51:29 +0000 Subject: [Freeswitch-users] Verto on mobile only connects in incognito mode Message-ID: Hi, Been having a bit of an issue with Verto on mobile connecting to a conference - It works fine when using incognito mode, but not when browsing normally (other than intermittently on the first attempt). The logs make it look like it's an ICE issue (line 544), but incognito mode working is pretty confusing - at least for me. I was wondering if anyone had any ideas on what the exact problem could be? I've attached the logs to this email, hope that's the right way to do things on the mailing list. Thanks, Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: incognito Type: application/octet-stream Size: 42607 bytes Desc: incognito URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: normal Type: application/octet-stream Size: 50688 bytes Desc: normal URL: From shaun.stokes at itec-support.co.uk Fri Sep 21 14:56:32 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 21 Sep 2018 14:56:32 +0000 Subject: [Freeswitch-users] How to set invite domain on a sofia bridge? In-Reply-To: References: Message-ID: Hi All, Freeswitch 1.8.1 Using sofia bridge inside a LUA script trying to set the domain on the INVITE to look like this: INVITE sip:220 at mydomain.com:5060 SIP/2.0 Instead our invite looks like this: INVITE sip:220 at 127.0.0.111:5060 SIP/2.0 We've tried setting various variables for domain\host, as follows: domain_name = "mydomain.com"; session:execute("bridge_export","sip_invite_domain="..domain_name); session:execute("bridge_export","domain_name="..domain_name); session:execute("bridge_export","sip_from_host="..domain_name); session:execute("bridge_export","sip_req_host="..domain_name); session:execute("bridge_export","sip_to_host="..domain_name); session:execute("bridge_export","sip_contact_host="..domain_name); session:execute("bridge_export","sip_via_host="..domain_name); session:execute("bridge", "sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried putting the above exported variables inside the bridge using {}, like so: session:execute("bridge", "{sip_invite_domain="..domain_name..",domain_name="..domain_name.."...etc...}sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried with-out overriding the To header, like so: session:execute("bridge", "sofia/local_internal/sip:"..dialed_extension.."@127.0.0.111:5060); How do we set the invite domain on a bridge? Hope someone here can help. Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From ba.lerest at gmail.com Fri Sep 21 15:16:45 2018 From: ba.lerest at gmail.com (Bastien LE REST) Date: Fri, 21 Sep 2018 17:16:45 +0200 Subject: [Freeswitch-users] video conference call, member audio only Message-ID: Hello, I’m trying to make a conference call. (A) bridge a video call to (B) and (B) answer, I transfer to the conference just before bridging using: api_before_bridge=uuid_transfer ${uuid} -both 'conference:TEST’ inline. Everything works fine. (B) bridge another audio call to (C) and (C) answer, I transfer to the conference using the same command: api_before_bridge=uuid_transfer ${uuid} -both 'conference:TEST’ inline. Audio is fine, I can hear (A), (B) and (C). The video between (A) and (B) stop and I’m getting the following warning on FS console: [WARNING] switch_core_media.c:14282 sofia/external/+33969321515 has no video codec. Does it possible to have 2 video members and 1 audio only member in the same conference? Here full traces: https://pastebin.com/SExxV6EC Thank you, -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Sep 21 15:29:08 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 21 Sep 2018 15:29:08 +0000 Subject: [Freeswitch-users] How to set invite domain on a sofia bridge? In-Reply-To: References: , Message-ID: We figured this out looking a previous thread with Ken, just need to set the fs_path on the bridge then we can modify the domain. Source: In FreeSWITCH if you want to send it to a proxy see fs_path… example ”/> ________________________________ From: Shaun Stokes Sent: 21 September 2018 15:56:32 To: FreeSWITCH Users Help Subject: How to set invite domain on a sofia bridge? Hi All, Freeswitch 1.8.1 Using sofia bridge inside a LUA script trying to set the domain on the INVITE to look like this: INVITE sip:220 at mydomain.com:5060 SIP/2.0 Instead our invite looks like this: INVITE sip:220 at 127.0.0.111:5060 SIP/2.0 We've tried setting various variables for domain\host, as follows: domain_name = "mydomain.com"; session:execute("bridge_export","sip_invite_domain="..domain_name); session:execute("bridge_export","domain_name="..domain_name); session:execute("bridge_export","sip_from_host="..domain_name); session:execute("bridge_export","sip_req_host="..domain_name); session:execute("bridge_export","sip_to_host="..domain_name); session:execute("bridge_export","sip_contact_host="..domain_name); session:execute("bridge_export","sip_via_host="..domain_name); session:execute("bridge", "sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried putting the above exported variables inside the bridge using {}, like so: session:execute("bridge", "{sip_invite_domain="..domain_name..",domain_name="..domain_name.."...etc...}sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried with-out overriding the To header, like so: session:execute("bridge", "sofia/local_internal/sip:"..dialed_extension.."@127.0.0.111:5060); How do we set the invite domain on a bridge? Hope someone here can help. Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Sep 21 20:45:22 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 21 Sep 2018 15:45:22 -0500 Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. In-Reply-To: <767b898f-532f-4618-c49e-6a97777fe413@mail.ru> References: <767b898f-532f-4618-c49e-6a97777fe413@mail.ru> Message-ID: Have a look here https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech On Fri, Sep 21, 2018 at 8:17 AM Ilya Spirin wrote: > Thanks for reply. > > Could you tell more please? I send a playback command already and file is > playing for some time. > But exactly detect_speech application ends quickly. > > Tried to send "detect_speech resume". There is still no effect (see > pastebin). > > On 21.09.2018 03:29, Anthony Minessale wrote: > > Its async so send the call to do something like park so it doesn't end. > > On Thu, Sep 20, 2018 at 4:21 PM Ilya Spirin via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: Ilya Spirin >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Thu, 20 Sep 2018 13:12:49 -0700 (MST) >> Subject: Detecting speech over ESL using mod_unimrcp. >> Hello everyone! >> >> I'm trying to recognize speech with my application. FS built with >> mod_unimrcp. Also I have installed UniMRCP Server with Google Speech >> Recognition plugin. >> >> play_and_detect_speech application works fine. But I have to stop playback >> manually that's why I'm sending some commands via ESL. >> >> Here is ESL dump: >> https://pastebin.com/p1jUrhn6 >> After sending answer, playback and detect_speech commands I immediately >> got >> "CHANNEL_EXECUTE_COMPLETE" event for detect_speech application. >> >> Please help. What I'm doing wrong? >> Thank you. >> >> -- >> Best regards >> Ilya Spirin >> >> >> >> -- >> Sent from: http://freeswitch-users.2379917.n2.nabble.com/ >> >> >> >> >> ---------- Forwarded message ---------- >> From: Ilya Spirin via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Thu, 20 Sep 2018 14:21:25 -0700 (PDT) >> Subject: [Freeswitch-users] Detecting speech over ESL using mod_unimrcp. >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Sat Sep 22 06:06:23 2018 From: me at nevian.org (Serge S. Yuriev) Date: Sat, 22 Sep 2018 09:06:23 +0300 Subject: [Freeswitch-users] How to set invite domain on a sofia bridge? In-Reply-To: References: , Message-ID: <4151381537596383@myt6-2fee75662a4f.qloud-c.yandex.net> An HTML attachment was scrubbed... URL: From alex at freeswitch.com Sun Sep 23 03:58:06 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Sun, 23 Sep 2018 12:58:06 +0900 Subject: [Freeswitch-users] Verto on mobile only connects in incognito mode In-Reply-To: References: Message-ID: Hi, Try to wipe all app data/reset it to factory defaults. Regards, Alex On Sat, Sep 22, 2018 at 3:56 AM Nick Woodward wrote: > Hi, > > Been having a bit of an issue with Verto on mobile connecting to a > conference - It works fine when using incognito mode, but not when browsing > normally (other than intermittently on the first attempt). The logs make it > look like it's an ICE issue (line 544), but incognito mode working is > pretty confusing - at least for me. > > I was wondering if anyone had any ideas on what the exact problem could > be? I've attached the logs to this email, hope that's the right way to do > things on the mailing list. > > Thanks, > > Nick > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Mon Sep 24 12:19:20 2018 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 24 Sep 2018 13:19:20 +0100 Subject: [Freeswitch-users] ring all devices other than self with registration id as same In-Reply-To: References: Message-ID: You could write and call a script that calls sofia_contact and filters out their device. On Wed, 19 Sep 2018 at 21:49, Bipin Patel wrote: > hi, > > i have a client registered to FS using same id on 2 devices and i created > a dialplan entry such that if he calls 000 then it can ring all devices > registered with same id, problem is the call also comes to himself on the > device he calls 000 from, is there a way to ring all other devices > registered other than his (multiple registration is enabled on profile) > > the bridge statement im using is as below currently where 5656 is his > registered userid > > > > > -- > Regards, > Bipin > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 24 15:36:47 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 24 Sep 2018 16:36:47 +0100 Subject: [Freeswitch-users] gateway/proxy/providers Message-ID: hello guys, I have a proxy in front of freeswitch. I want freeswitch to use the proxy for everything including the termination to my provider. If I configured the gateway like this: The registration works properly, the REGISTER goes to the proxy like REGISTER sip:myprovider.com;transport=udp SIP/2.0 and the proxy forwards it fine, and registration works great. But when i try to call via this gateway, the call goes STRAIGHT to the provider! If, on the other hand, i configure my gateway as: ** Registration does NOT work, because the uri is: REGISTER sip:*myproxy.domain.com *;transport=udp SIP/2.0 Because there it no registration on that proxy! Please help me with this! David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 24 16:14:50 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 24 Sep 2018 17:14:50 +0100 Subject: [Freeswitch-users] gateway/proxy/provider Message-ID: Hello guys, I have a proxy in front of freeswitch. I want freeswitch to use the proxy for everything including the termination to my provider. If I configured the gateway like this: The registration works properly, the REGISTER goes to the proxy like REGISTER sip:myprovider.com;transport=udp SIP/2.0 and the proxy forwards it fine, and registration works great. But when i try to call via this gateway, the call goes STRAIGHT to the provider! If, on the other hand, i configure my gateway as: ** Registration does NOT work, because the uri is: REGISTER sip:*myproxy.domain.com *;transport=udp SIP/2.0 Because there it no registration on that proxy! Please help me with this! ReplyForward ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Mon Sep 24 19:27:14 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 25 Sep 2018 00:27:14 +0500 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] Message-ID: Hey Users, I am using 8 FreeSWITCH servers and all of them are on Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit OS: Debian 8.10 64Bit I am using mod_event socket to dial the calls and its around 20Calls per second, and Max to 200 Concurrent calls. But my FreeSWITCH server crashes randomly, I enabled the debug logs to see if i could find something(log set for production was warning, but i did that debug to see if i could find something). I can see following logs on crash(service is configured to restart on crash), but getting no clue why its crashing https://pastebin.com/Xga5Fh70 P.S: all of the 8 servers are crashing on random times. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 24 20:12:35 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 24 Sep 2018 21:12:35 +0100 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] In-Reply-To: References: Message-ID: You're best bet is to enable debugging flags and get a core dump and ask the guys at freeswitch for help. On Mon, Sep 24, 2018, 20:56 Bilal Abbasi wrote: > Hey Users, > I am using 8 FreeSWITCH servers and all of them are on > Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit > OS: Debian 8.10 64Bit > > I am using mod_event socket to dial the calls and its around 20Calls per > second, and Max to 200 Concurrent calls. > But my FreeSWITCH server crashes randomly, I enabled the debug logs to see > if i could find something(log set for production was warning, but i did > that debug to see if i could find something). > I can see following logs on crash(service is configured to restart on > crash), but getting no clue why its crashing > https://pastebin.com/Xga5Fh70 > > P.S: all of the 8 servers are crashing on random times. > > Regards > Abbasi > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Mon Sep 24 23:15:22 2018 From: brians at iptel.co (Brian :) Date: Tue, 25 Sep 2018 00:15:22 +0100 Subject: [Freeswitch-users] gateway/proxy/provider In-Reply-To: References: Message-ID: Hi David, Have you tried using fs_path for your ob call? On Monday, September 24, 2018, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello guys, > > I have a proxy in front of freeswitch. I want freeswitch to use the proxy for everything including the termination to my provider. > If I configured the gateway like this: > > > > > > > > > > > > > The registration works properly, the REGISTER goes to the proxy like > REGISTER sip:myprovider.com;transport=udp SIP/2.0 > > and the proxy forwards it fine, and registration works great. > But when i try to call via this gateway, the call goes STRAIGHT to the provider! > > If, on the other hand, i configure my gateway as: > > > > > > > > > > > > > > Registration does NOT work, because the uri is: > REGISTER sip:myproxy.domain.com;transport=udp SIP/2.0 > > Because there it no registration on that proxy! > > Please help me with this! > < https://ci3.googleusercontent.com/proxy/brwtshoO8pFoWzivuTYTqzcgbWbRGn8NoU8wERg_KJm3_dQfWsYZcw5SQWdMXaW39zaYym3eYNHeLECuyKxBYYGgtAEK-2_yIIyzW_3fVstyQg=s0-d-e1-ft#https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif > > < https://ci4.googleusercontent.com/proxy/9clsM5l2LCOkWgFnmtC5AcB373zu9V-tEEyatn76dk2ZceD21xhzpoBQMEobe-KTxvgDc8KEuHnXehdXXisA0SyTJTqhqPL0ruSDkX2o0a6L3s6jJhTyQg8sFFIO962ry92wcJM4Kwsd_M35Q0XkRp-HD-K8lZ3vsFRjtzpSzpUzLsBmnt7faXVA_lFxntideboPttRHivl243_NxbSfskJ5CZreGxDn_D7UM22O9DyJLt05oVApcWW40QLN8mf3Bk9teekTmQDUKrTp47yV=s0-d-e1-ft#https://plus.google.com/u/1/_/focus/photos/public/AIbEiAIAAABDCJauh-KD253CPiILdmNhcmRfcGhvdG8qKGU2NzQzMmZmZDJiNjdjMzMxNmU5OTUwY2QyNDM4ZTZkOGMwYWVjOGUwAWX3K85GhrhrlY4u9DzEPdh2yDdT?sz=32 > > ReplyForward > < https://ci3.googleusercontent.com/proxy/QOKnIbfXiMGPEATDM-zVHLnvbr-59bIJZkjc7OCBdzZSIRNEqN_UNL-69eQbQteFagJkLymInNYvYDr9DSdJN8tR2MNDYH2uIXiBs96-ksex8kKH0AsYoCmiPzmAytGP6dakFh7FHPimz27nUt-UAJniOvS-Ap0FyIVhwAqljzfJWMN47Q2PsufpvFj2EZwVZHV6QMAdJmA3EvyMxBUUifozgg=s0-d-e1-ft#https://mailfoogae.appspot.com/t?sender=aZGF2aWQudmlsbGFzbWlsLndvcmtAZ21haWwuY29t&type=zerocontent&guid=5141107a-6fb2-4634-b5c4-439c01c97e30>ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Sep 24 23:41:27 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 25 Sep 2018 00:41:27 +0100 Subject: [Freeswitch-users] gateway/proxy/provider In-Reply-To: References: Message-ID: Any help on this guys? On Mon, Sep 24, 2018, 17:14 David Villasmil wrote: > Hello guys, > > > I have a proxy in front of freeswitch. I want freeswitch to use the proxy > for everything including the termination to my provider. > > If I configured the gateway like this: > > > > > > > > > > > > > > > The registration works properly, the REGISTER goes to the proxy like > > REGISTER sip:myprovider.com;transport=udp SIP/2.0 > > and the proxy forwards it fine, and registration works great. > But when i try to call via this gateway, the call goes STRAIGHT to the > provider! > > If, on the other hand, i configure my gateway as: > > > > > > > > > > > > ** > > > > Registration does NOT work, because the uri is: > > REGISTER sip:*myproxy.domain.com *;transport=udp > SIP/2.0 > > Because there it no registration on that proxy! > > > Please help me with this! > ReplyForward > ᐧ > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Tue Sep 25 02:22:37 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 24 Sep 2018 19:22:37 -0700 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] In-Reply-To: References: Message-ID: And most likely first request will be open a JIRA ticket and test on latest master in case it’s fixed already. :) On Mon, Sep 24, 2018 at 14:45 David Villasmil < david.villasmil.work at gmail.com> wrote: > You're best bet is to enable debugging flags and get a core dump and ask > the guys at freeswitch for help. > > On Mon, Sep 24, 2018, 20:56 Bilal Abbasi wrote: > >> Hey Users, >> I am using 8 FreeSWITCH servers and all of them are on >> Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit >> OS: Debian 8.10 64Bit >> >> I am using mod_event socket to dial the calls and its around 20Calls per >> second, and Max to 200 Concurrent calls. >> But my FreeSWITCH server crashes randomly, I enabled the debug logs to >> see if i could find something(log set for production was warning, but i did >> that debug to see if i could find something). >> I can see following logs on crash(service is configured to restart on >> crash), but getting no clue why its crashing >> https://pastebin.com/Xga5Fh70 >> >> P.S: all of the 8 servers are crashing on random times. >> >> Regards >> Abbasi >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Tue Sep 25 02:24:09 2018 From: joel at textplus.com (Joel Serrano) Date: Mon, 24 Sep 2018 19:24:09 -0700 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304 On Mon, Sep 24, 2018 at 19:22 Joel Serrano wrote: > And most likely first request will be open a JIRA ticket and test on > latest master in case it’s fixed already. :) > > > On Mon, Sep 24, 2018 at 14:45 David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> You're best bet is to enable debugging flags and get a core dump and ask >> the guys at freeswitch for help. >> >> On Mon, Sep 24, 2018, 20:56 Bilal Abbasi wrote: >> >>> Hey Users, >>> I am using 8 FreeSWITCH servers and all of them are on >>> Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit >>> OS: Debian 8.10 64Bit >>> >>> I am using mod_event socket to dial the calls and its around 20Calls per >>> second, and Max to 200 Concurrent calls. >>> But my FreeSWITCH server crashes randomly, I enabled the debug logs to >>> see if i could find something(log set for production was warning, but i did >>> that debug to see if i could find something). >>> I can see following logs on crash(service is configured to restart on >>> crash), but getting no clue why its crashing >>> https://pastebin.com/Xga5Fh70 >>> >>> P.S: all of the 8 servers are crashing on random times. >>> >>> Regards >>> Abbasi >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Sep 25 14:44:20 2018 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 25 Sep 2018 18:44:20 +0400 Subject: [Freeswitch-users] ring all devices other than self with registration id as same In-Reply-To: References: Message-ID: <078392b3-9fd9-e104-fec1-77f40658e148@xbipin.com> hi, after day of readings docs etc i wasnt able to find anything useful so ended up writing a lua script (first time working with lua), i had figured by then there was no other way without writing a script. below is the script if it helps any1 else, its not perfect but works. I pass the calling party's username and his contact ip:port function string:split(delimiter)     local result = { }     local from = 1     local delim_from, delim_to = string.find( self, delimiter, from )     while delim_from do         table.insert( result, string.sub( self, from , delim_from-1 ) )         from = delim_to + 1         delim_from, delim_to = string.find( self, delimiter, from )     end     table.insert( result, string.sub( self, from ) )     return result end function isempty(s)     return s == nil or s == '' end vps_id = argv[1] vps_ip = argv[2] check_ip = vps_ip:match("(%d+%.%d+%.%d+%.%d+%:%d+)") api = freeswitch.API(); result = api:execute("sofia_contact",vps_id); if (result == "error/user_not_registered") then     session:execute("bridge","error/user_not_registered"); else     t = {}     array_list = string.split(result,",")     if (#(array_list) > 1) then         for x,y in pairs(array_list) do             if isempty(string.find(y,check_ip)) then                 t[#t+1] = tostring(y)             end         end         out_myvar = table.concat(t,",")     else         for x,y in pairs(array_list) do             if isempty(string.find(y,check_ip)) then                 out_myvar = tostring(y)             end         end     end     if isempty(out_myvar) then         session:execute("bridge","error/user_not_registered");     else         session:execute("bridge", out_myvar);     end end Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] ring all devices other than self with registration id as same From: Steven Ayre To: FreeSWITCH Users Help Date: 9/24/2018, 4:19:20 PM > You could write and call a script that calls sofia_contact and filters > out their device. > > On Wed, 19 Sep 2018 at 21:49, Bipin Patel > wrote: > > hi, > > i have a client registered to FS using same id on 2 devices and i > created a dialplan entry such that if he calls 000 then it can > ring all devices registered with same id, problem is the call also > comes to himself on the device he calls 000 from, is there a way > to ring all other devices registered other than his (multiple > registration is enabled on profile) > > the bridge statement im using is as below currently where 5656 is > his registered userid > > > > > -- > Regards, > Bipin > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaduww at gmail.com Tue Sep 25 15:25:27 2018 From: kaduww at gmail.com (Carlos Eduardo) Date: Tue, 25 Sep 2018 12:25:27 -0300 Subject: [Freeswitch-users] gateway/proxy/providers In-Reply-To: References: Message-ID: Hello, Try using It worked for me. 2018-09-24 12:36 GMT-03:00 David Villasmil : > hello guys, > > I have a proxy in front of freeswitch. I want freeswitch to use the proxy > for everything including the termination to my provider. > > If I configured the gateway like this: > > > > > > > > > > > > > > > The registration works properly, the REGISTER goes to the proxy like > > REGISTER sip:myprovider.com;transport=udp SIP/2.0 > > and the proxy forwards it fine, and registration works great. > But when i try to call via this gateway, the call goes STRAIGHT to the > provider! > > If, on the other hand, i configure my gateway as: > > > > > > > > > > > > ** > > > > Registration does NOT work, because the uri is: > > REGISTER sip:*myproxy.domain.com *;transport=udp > SIP/2.0 > > Because there it no registration on that proxy! > > > Please help me with this! > > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > ᐧ > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, OCP, dCAA* *E-mail:* *kaduww at gmail.com * *Fone:* +55 48 9981-0894 *Skype:* carlos.e.wagner www.blogdovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragic.dusan at gmail.com Tue Sep 25 15:39:48 2018 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Tue, 25 Sep 2018 17:39:48 +0200 Subject: [Freeswitch-users] gateway/proxy/provider In-Reply-To: References: Message-ID: Try with outbound-proxy param instead of proxy. On Mon, Sep 24, 2018, 22:06 David Villasmil wrote: > Hello guys, > > > I have a proxy in front of freeswitch. I want freeswitch to use the proxy > for everything including the termination to my provider. > > If I configured the gateway like this: > > > > > > > > > > > > > > > The registration works properly, the REGISTER goes to the proxy like > > REGISTER sip:myprovider.com;transport=udp SIP/2.0 > > and the proxy forwards it fine, and registration works great. > But when i try to call via this gateway, the call goes STRAIGHT to the > provider! > > If, on the other hand, i configure my gateway as: > > > > > > > > > > > > ** > > > > Registration does NOT work, because the uri is: > > REGISTER sip:*myproxy.domain.com *;transport=udp > SIP/2.0 > > Because there it no registration on that proxy! > > > Please help me with this! > ReplyForward > ᐧ > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Tue Sep 25 16:25:49 2018 From: joel at textplus.com (Joel Serrano) Date: Tue, 25 Sep 2018 09:25:49 -0700 Subject: [Freeswitch-users] Encrypted video calls demo Message-ID: Hi All, I friend recently asked me about a PoC he is setting up and he needs encrypted video calls..blablabla. Long story short, I suggested to give FS + Verto a try blablabla.. but I wasn't sure if the "easy getting started" *today* would be debian8+fs16 or debian9+fs18... So my question is, for a really quick demo of FS+Verto would you go with 1.6 and debian8 or 1.8 and debian9 (for a first-time-fs-user)? I personally would test with v1.8 but not sure if the installation and can still be a little more tricky than v1.6 on debian8...? Opinions?? :D Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Tue Sep 25 16:28:20 2018 From: social at bohboh.info (Social Boh) Date: Tue, 25 Sep 2018 11:28:20 -0500 Subject: [Freeswitch-users] NibbleBill not Hangup Message-ID: <2ecd2501-502a-f56d-8084-937e6c7f2fa9@bohboh.info> Hello, I have a problem with NibbleBill module. If a user have 0 or less credit, FS not hangup the call: [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account 1000 (balance = -0.003100) [DEBUG] mod_nibblebill.c:523 Comparing -0.003100 to hangup balance of 0.000000, taking into account minimum charge of 0.000000 [DEBUG] mod_nibblebill.c:526 Balance of -0.003100 fell below allowed amount of 0.000000! (Account 1000) this is only appear at the end of the call. In the configuration module I'm using:     hangup extensión:             dialplan:       Thank you Regards -- --- I'm SoCIaL, MayBe From steveayre at gmail.com Tue Sep 25 18:46:05 2018 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 25 Sep 2018 19:46:05 +0100 Subject: [Freeswitch-users] ring all devices other than self with registration id as same In-Reply-To: <078392b3-9fd9-e104-fec1-77f40658e148@xbipin.com> References: <078392b3-9fd9-e104-fec1-77f40658e148@xbipin.com> Message-ID: Thanks for sharing the script! Always helps avoid https://xkcd.com/979/ On Tue, 25 Sep 2018 at 15:44, Bipin Patel wrote: > hi, > > after day of readings docs etc i wasnt able to find anything useful so > ended up writing a lua script (first time working with lua), i had figured > by then there was no other way without writing a script. > > below is the script if it helps any1 else, its not perfect but works. I > pass the calling party's username and his contact ip:port > > function string:split(delimiter) > local result = { } > local from = 1 > local delim_from, delim_to = string.find( self, delimiter, from ) > while delim_from do > table.insert( result, string.sub( self, from , delim_from-1 ) ) > from = delim_to + 1 > delim_from, delim_to = string.find( self, delimiter, from ) > end > table.insert( result, string.sub( self, from ) ) > return result > end > > function isempty(s) > return s == nil or s == '' > end > > vps_id = argv[1] > vps_ip = argv[2] > check_ip = vps_ip:match("(%d+%.%d+%.%d+%.%d+%:%d+)") > api = freeswitch.API(); > result = api:execute("sofia_contact",vps_id); > if (result == "error/user_not_registered") then > session:execute("bridge","error/user_not_registered"); > else > t = {} > array_list = string.split(result,",") > if (#(array_list) > 1) then > for x,y in pairs(array_list) do > if isempty(string.find(y,check_ip)) then > t[#t+1] = tostring(y) > end > end > out_myvar = table.concat(t,",") > else > for x,y in pairs(array_list) do > if isempty(string.find(y,check_ip)) then > out_myvar = tostring(y) > end > end > end > if isempty(out_myvar) then > session:execute("bridge","error/user_not_registered"); > else > session:execute("bridge", out_myvar); > end > end > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] ring all devices other than self with > registration id as same > From: Steven Ayre > To: FreeSWITCH Users Help > > Date: 9/24/2018, 4:19:20 PM > > You could write and call a script that calls sofia_contact and filters out > their device. > > On Wed, 19 Sep 2018 at 21:49, Bipin Patel wrote: > >> hi, >> >> i have a client registered to FS using same id on 2 devices and i created >> a dialplan entry such that if he calls 000 then it can ring all devices >> registered with same id, problem is the call also comes to himself on the >> device he calls 000 from, is there a way to ring all other devices >> registered other than his (multiple registration is enabled on profile) >> >> the bridge statement im using is as below currently where 5656 is his >> registered userid >> >> >> >> >> -- >> Regards, >> Bipin >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Wed Sep 26 07:04:54 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 26 Sep 2018 07:04:54 +0000 Subject: [Freeswitch-users] How to set invite domain on a sofia bridge? In-Reply-To: <4151381537596383@myt6-2fee75662a4f.qloud-c.yandex.net> References: , , <4151381537596383@myt6-2fee75662a4f.qloud-c.yandex.net> Message-ID: I've not tested 'fs_path' on a gateway, if you're using a gateway which is configured on FreeSWITCH then I believe you can specify the domain and IP address (proxy) in the gateway config. You should only need to use 'fs_path' when attempting to bridge directly with the destination with-out going through either a gateway or user configured on FreeSWITCH. ________________________________ From: FreeSWITCH-users on behalf of Serge S.Yuriev Sent: 22 September 2018 07:06:23 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to set invite domain on a sofia bridge? Hi Is this compatible with gateway and distributor? We have pool of "proxies" (gws) to terminate to. Also can we set from-domain per gw config? -- Wbr, Serge via mobile 21.09.2018, 21:41, "Shaun Stokes" : We figured this out looking a previous thread with Ken, just need to set the fs_path on the bridge then we can modify the domain. Source: In FreeSWITCH if you want to send it to a proxy see fs_path… example ”/> ________________________________ From: Shaun Stokes Sent: 21 September 2018 15:56:32 To: FreeSWITCH Users Help Subject: How to set invite domain on a sofia bridge? Hi All, Freeswitch 1.8.1 Using sofia bridge inside a LUA script trying to set the domain on the INVITE to look like this: INVITE sip:220 at mydomain.com:5060 SIP/2.0 Instead our invite looks like this: INVITE sip:220 at 127.0.0.111:5060 SIP/2.0 We've tried setting various variables for domain\host, as follows: domain_name = "mydomain.com"; session:execute("bridge_export","sip_invite_domain="..domain_name); session:execute("bridge_export","domain_name="..domain_name); session:execute("bridge_export","sip_from_host="..domain_name); session:execute("bridge_export","sip_req_host="..domain_name); session:execute("bridge_export","sip_to_host="..domain_name); session:execute("bridge_export","sip_contact_host="..domain_name); session:execute("bridge_export","sip_via_host="..domain_name); session:execute("bridge", "sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried putting the above exported variables inside the bridge using {}, like so: session:execute("bridge", "{sip_invite_domain="..domain_name..",domain_name="..domain_name.."...etc...}sofia/internal/sip:"..dialed_extension.."@127.0.0.111:5060^"..dialed_extension.."@"..domain_name"); Tried with-out overriding the To header, like so: session:execute("bridge", "sofia/local_internal/sip:"..dialed_extension.."@127.0.0.111:5060); How do we set the invite domain on a bridge? Hope someone here can help. Thanks, Shaun _________________________________________________________________________ Professional FreeSWITCH Services sales at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Wed Sep 26 09:51:50 2018 From: infos at madovsky.org (Madovsky) Date: Wed, 26 Sep 2018 02:51:50 -0700 Subject: [Freeswitch-users] NibbleBill not Hangup In-Reply-To: <2ecd2501-502a-f56d-8084-937e6c7f2fa9@bohboh.info> References: <2ecd2501-502a-f56d-8084-937e6c7f2fa9@bohboh.info> Message-ID: <64dad416-33a2-fb95-359f-d94bd49b11c5@madovsky.org> Do you have any extension named "hangup" in your default dialplan? On 9/25/2018 9:28 AM, Social Boh wrote: > Hello, > > I have a problem with NibbleBill module. > > If a user have 0 or less credit, FS not hangup the call: > > [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account > 1000 (balance = -0.003100) > [DEBUG] mod_nibblebill.c:523 Comparing -0.003100 to hangup balance of > 0.000000, taking into account minimum charge of 0.000000 > [DEBUG] mod_nibblebill.c:526 Balance of -0.003100 fell below allowed > amount of 0.000000! (Account 1000) > > this is only appear at the end of the call. > > In the configuration module I'm using: > > > > >     > > hangup extensión: > > > >   >     >     >   > > > > dialplan: > > >       > > Thank you > > Regards > > From bilaln018 at gmail.com Wed Sep 26 11:45:34 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 26 Sep 2018 16:45:34 +0500 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] In-Reply-To: References: Message-ID: Thanks for your time, i just open the JIRA ticket. https://freeswitch.org/jira/browse/FS-11411 Regards Abbasi On Wed, Sep 26, 2018 at 3:56 PM Joel Serrano wrote: > > https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304 > > > On Mon, Sep 24, 2018 at 19:22 Joel Serrano wrote: > >> And most likely first request will be open a JIRA ticket and test on >> latest master in case it’s fixed already. :) >> >> >> On Mon, Sep 24, 2018 at 14:45 David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> You're best bet is to enable debugging flags and get a core dump and ask >>> the guys at freeswitch for help. >>> >>> On Mon, Sep 24, 2018, 20:56 Bilal Abbasi wrote: >>> >>>> Hey Users, >>>> I am using 8 FreeSWITCH servers and all of them are on >>>> Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit >>>> OS: Debian 8.10 64Bit >>>> >>>> I am using mod_event socket to dial the calls and its around 20Calls >>>> per second, and Max to 200 Concurrent calls. >>>> But my FreeSWITCH server crashes randomly, I enabled the debug logs to >>>> see if i could find something(log set for production was warning, but i did >>>> that debug to see if i could find something). >>>> I can see following logs on crash(service is configured to restart on >>>> crash), but getting no clue why its crashing >>>> https://pastebin.com/Xga5Fh70 >>>> >>>> P.S: all of the 8 servers are crashing on random times. >>>> >>>> Regards >>>> Abbasi >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From carsten at ng-voice.com Wed Sep 26 12:31:10 2018 From: carsten at ng-voice.com (Carsten Bock) Date: Wed, 26 Sep 2018 14:31:10 +0200 Subject: [Freeswitch-users] Video Upgrade/Downgrade and Transcoding Message-ID: Hi, I am using FreeSwitch Release: FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451 2018-08-31 19:05:39Z 64bit) is ready I have the following, rather simple scenario, where I can't figure out, how to configure it properly. I use FreeSwitch as Transcoding B2B-UA/SBC. All the logic is done on a Kamailio in Front of FreeSwitch. Thus my dialplan is rather simple: In order to provide transcoding, I have set the following in my internal.xml profile: In vars.xml, I have the following: This works like a charm for all Voice-Calls, the B-Party gets an offer including all the codecs from the list, e.g. A-Party supports only G722, B-Party supports only PCMU, so FreeSwitch does the transcoding. However, it leads to issues, when it comes to Video. As soon as I add H264 and VP8 to the codecs list (global_codec_pref / outbound_codec_prefs), the call to my B-Party includes the offer for Video as well, even if the A-Party did not offer Video initially. Is there an easy way to solve this? What are the right settings for such scenario? I tried the following in my dialplan: This would remove the Video from the initial invite, if the A-Party did not offer Video initially. However, if I then enable Video at a later stage (Re-INVITE with m=video), FreeSwitch will not forward the new stream to the B-Party and then instead just send a re-INVITE with Audio only to the B-Party. Where did I go wrong? Any ideas? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten at ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ From david.villasmil.work at gmail.com Wed Sep 26 13:52:30 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 26 Sep 2018 14:52:30 +0100 Subject: [Freeswitch-users] gateway/proxy/providers In-Reply-To: References: Message-ID: Hello Carlos, Thanks, i'll try that David On Wed, Sep 26, 2018, 11:23 Carlos Eduardo wrote: > Hello, > > Try using > > It worked for me. > > > 2018-09-24 12:36 GMT-03:00 David Villasmil >: > >> hello guys, >> >> I have a proxy in front of freeswitch. I want freeswitch to use the proxy >> for everything including the termination to my provider. >> >> If I configured the gateway like this: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The registration works properly, the REGISTER goes to the proxy like >> >> REGISTER sip:myprovider.com;transport=udp SIP/2.0 >> >> and the proxy forwards it fine, and registration works great. >> But when i try to call via this gateway, the call goes STRAIGHT to the >> provider! >> >> If, on the other hand, i configure my gateway as: >> >> >> >> >> >> >> >> >> >> >> >> ** >> >> >> >> Registration does NOT work, because the uri is: >> >> REGISTER sip:*myproxy.domain.com *;transport=udp >> SIP/2.0 >> >> Because there it no registration on that proxy! >> >> >> Please help me with this! >> >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> ᐧ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > *Carlos E. Wagner* > *Tecnólogo em Telecomunicações, OCP, dCAA* > > *E-mail:* *kaduww at gmail.com * > *Fone:* +55 48 9981-0894 > *Skype:* carlos.e.wagner > www.blogdovoip.com > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Sep 26 13:58:51 2018 From: brian at freeswitch.com (Brian West) Date: Wed, 26 Sep 2018 08:58:51 -0500 Subject: [Freeswitch-users] [FreeSWITCH Servers Crashes][Unexpectedly] In-Reply-To: References: Message-ID: In this case, you're executing 'fsctl crash', so I'd review what might be calling that. /b On Wed, Sep 26, 2018 at 7:56 AM Bilal Abbasi wrote: > Thanks for your time, i just open the JIRA ticket. > https://freeswitch.org/jira/browse/FS-11411 > > Regards > Abbasi > > On Wed, Sep 26, 2018 at 3:56 PM Joel Serrano wrote: > >> >> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304 >> >> >> On Mon, Sep 24, 2018 at 19:22 Joel Serrano wrote: >> >>> And most likely first request will be open a JIRA ticket and test on >>> latest master in case it’s fixed already. :) >>> >>> >>> On Mon, Sep 24, 2018 at 14:45 David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> You're best bet is to enable debugging flags and get a core dump and >>>> ask the guys at freeswitch for help. >>>> >>>> On Mon, Sep 24, 2018, 20:56 Bilal Abbasi wrote: >>>> >>>>> Hey Users, >>>>> I am using 8 FreeSWITCH servers and all of them are on >>>>> Version 1.6.20 git 43a9feb 2018-05-07 18:56:11Z 64bit >>>>> OS: Debian 8.10 64Bit >>>>> >>>>> I am using mod_event socket to dial the calls and its around 20Calls >>>>> per second, and Max to 200 Concurrent calls. >>>>> But my FreeSWITCH server crashes randomly, I enabled the debug logs to >>>>> see if i could find something(log set for production was warning, but i did >>>>> that debug to see if i could find something). >>>>> I can see following logs on crash(service is configured to restart on >>>>> crash), but getting no clue why its crashing >>>>> https://pastebin.com/Xga5Fh70 >>>>> >>>>> P.S: all of the 8 servers are crashing on random times. >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Services >>>>> sales at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Wed Sep 26 14:00:32 2018 From: social at bohboh.info (Social Boh) Date: Wed, 26 Sep 2018 09:00:32 -0500 Subject: [Freeswitch-users] NibbleBill not Hangup In-Reply-To: <64dad416-33a2-fb95-359f-d94bd49b11c5@madovsky.org> References: <2ecd2501-502a-f56d-8084-937e6c7f2fa9@bohboh.info> <64dad416-33a2-fb95-359f-d94bd49b11c5@madovsky.org> Message-ID: Hello my mistake. Thank you for answer. regards --- I'm SoCIaL, MayBe El 26/09/2018 a las 04:51, Madovsky escribió: > Do you have any extension named "hangup" in your default dialplan? > > On 9/25/2018 9:28 AM, Social Boh wrote: >> Hello, >> >> I have a problem with NibbleBill module. >> >> If a user have 0 or less credit, FS not hangup the call: >> >> [DEBUG] mod_nibblebill.c:430 Retrieved current balance for account >> 1000 (balance = -0.003100) >> [DEBUG] mod_nibblebill.c:523 Comparing -0.003100 to hangup balance of >> 0.000000, taking into account minimum charge of 0.000000 >> [DEBUG] mod_nibblebill.c:526 Balance of -0.003100 fell below allowed >> amount of 0.000000! (Account 1000) >> >> this is only appear at the end of the call. >> >> In the configuration module I'm using: >> >> >> >> >>     >> >> hangup extensión: >> >> >> >>   >>     >>     >>   >> >> >> >> dialplan: >> >> >>       >> >> Thank you >> >> Regards >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > From vipkilla at gmail.com Wed Sep 26 15:10:32 2018 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 26 Sep 2018 11:10:32 -0400 Subject: [Freeswitch-users] h264 packetization modes Message-ID: is there a way to force h264 packetization mode 0 or 1 in freeswitch? running into an issue where a-leg h264 uses mode 1 but freeswitch sends mode 0 to b-leg. I've even tried disabling transcoding, and sending exact same A leg codec string to b-leg, and still, freeswitch uses different packetization mode. Any advice would be appreciated! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.terrasson at gmail.com Wed Sep 26 15:58:50 2018 From: julien.terrasson at gmail.com (Julien Terrasson) Date: Wed, 26 Sep 2018 17:58:50 +0200 Subject: [Freeswitch-users] Freeswitch 1.8 : Debian package : Is postgresql still the default core-dsn ? Message-ID: Hi all, I've been deploying freeswitch 1.8 package on a debian 9. When trying to connect to my own external postgres database i get this message : "2018-09-26 15:43:52.932465 [CRIT] switch_core_sqldb.c:468 Failure! PGSQL NOT AVAILABLE! Can't connect to DSN dbname BlabBlaBla.." Does this mean freeswitch 1.8 on debian 9 is built with a non postgresql core-dsn database ? Is there a way to query the core-dsn database type from freeswitch cli ? Thank for your help. J.Terrasson -------------- next part -------------- An HTML attachment was scrubbed... URL: From tiagoggsouza at gmail.com Wed Sep 26 16:12:13 2018 From: tiagoggsouza at gmail.com (=?UTF-8?Q?Tiago_Galv=C3=A3o_Gomes_de_Souza?=) Date: Wed, 26 Sep 2018 13:12:13 -0300 Subject: [Freeswitch-users] Fwd: Delivery Status Notification (Failure) In-Reply-To: <5babae2d.1c69fb81.989a8.20a3.GMR@mx.google.com> References: CAD5Lzev5q981NOzko05mLB1X3Zu3uUzQQtCDnN7Y+vSb_rPnhA@mail.gmail.com <5babae2d.1c69fb81.989a8.20a3.GMR@mx.google.com> Message-ID: someone can help? I'm trying to install freeswitch on centos 7 , follow the instructions : https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7#CentOS7andRHEL7-CentOS7andR. .. when a use a command "make" to start installation a receveid error like this image below: [image: freeswitch.JPG] -- Atenciosamente, Tiago Galvão Gomes de Souza. -- Atenciosamente, Tiago Galvão Gomes de Souza. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: icon.png Type: image/png Size: 1450 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.JPG Type: image/jpeg Size: 115739 bytes Desc: not available URL: From social at bohboh.info Wed Sep 26 17:47:00 2018 From: social at bohboh.info (Social Boh) Date: Wed, 26 Sep 2018 12:47:00 -0500 Subject: [Freeswitch-users] Fwd: Delivery Status Notification (Failure) In-Reply-To: References: <5babae2d.1c69fb81.989a8.20a3.GMR@mx.google.com> Message-ID: <7c90696a-6762-4e92-5287-b7ef44185b0e@bohboh.info> Hello, I'm using this one: https://www.voztovoice.org/?q=node/2381 maybe work for you too Regards --- I'm SoCIaL, MayBe El 26/09/2018 a las 11:12, Tiago Galvão Gomes de Souza escribió: > > someone can help? I'm trying to install freeswitch on centos 7 , > follow the instructions : > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7#CentOS7andRHEL7-CentOS7andR... > > when a use a command "make" to start installation a receveid error > like this image below: > > freeswitch.JPG > > -- > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > > -- > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.JPG Type: image/jpeg Size: 115739 bytes Desc: not available URL: From lists at telefaks.de Wed Sep 26 20:23:08 2018 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 26 Sep 2018 22:23:08 +0200 Subject: [Freeswitch-users] Store incomplete Fax pages? Message-ID: Hello, I discovered that, when a fax is partially received, only complete pages are saved (e.g. only 1st page out of 3 when fax stops in the middle of page 2). Is there a way, that Freeswitch also saves incomplete pages as TIF (e.g. 1st page and first half of page 2)? Thanks Peter From alex at freeswitch.com Thu Sep 27 02:52:18 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 27 Sep 2018 11:52:18 +0900 Subject: [Freeswitch-users] Freeswitch 1.8 : Debian package : Is postgresql still the default core-dsn ? In-Reply-To: References: Message-ID: Hi, Try to update to 1.8.2. Regards, Alex On Thu, Sep 27, 2018, 2:16 AM Julien Terrasson wrote: > Hi all, > > I've been deploying freeswitch 1.8 package on a debian 9. > When trying to connect to my own external postgres database i get this > message : > "2018-09-26 15:43:52.932465 [CRIT] switch_core_sqldb.c:468 Failure! PGSQL > NOT AVAILABLE! Can't connect to DSN dbname BlabBlaBla.." > Does this mean freeswitch 1.8 on debian 9 is built with a non postgresql > core-dsn database ? > Is there a way to query the core-dsn database type from freeswitch cli ? > Thank for your help. > > J.Terrasson > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Sep 27 03:41:16 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 27 Sep 2018 06:41:16 +0300 Subject: [Freeswitch-users] Fwd: Delivery Status Notification (Failure) In-Reply-To: <7c90696a-6762-4e92-5287-b7ef44185b0e@bohboh.info> References: <5babae2d.1c69fb81.989a8.20a3.GMR@mx.google.com> <7c90696a-6762-4e92-5287-b7ef44185b0e@bohboh.info> Message-ID: You can 1) disable sctp module; 2) remove strings with initmsg variable as not used. Sergey ср, 26 сент. 2018 г. в 21:53, Social Boh : > Hello, > > I'm using this one: > > https://www.voztovoice.org/?q=node/2381 > > maybe work for you too > > Regards > > --- > I'm SoCIaL, MayBe > > El 26/09/2018 a las 11:12, Tiago Galvão Gomes de Souza escribió: > > > someone can help? I'm trying to install freeswitch on centos 7 , follow > the instructions : > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7#CentOS7andRHEL7-CentOS7andR. > .. > > when a use a command "make" to start installation a receveid error like > this image below: > > [image: freeswitch.JPG] > > -- > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > > -- > Atenciosamente, > > Tiago Galvão Gomes de Souza. > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.JPG Type: image/jpeg Size: 115739 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.JPG Type: image/jpeg Size: 115739 bytes Desc: not available URL: From joel at textplus.com Thu Sep 27 09:02:30 2018 From: joel at textplus.com (Joel Serrano) Date: Thu, 27 Sep 2018 11:02:30 +0200 Subject: [Freeswitch-users] Video Upgrade/Downgrade and Transcoding In-Reply-To: References: Message-ID: Hi Carsten, I have a setup exactly like that... I’ll get you the combination of settings I have for FS to handle the codecs. As for your second problem, I can tell you it will not work, I have a quote from FS devs to implement the reinvite with video but as of right now you cannot switch from audio to audio+video (unless you proxy_media which of course you don’t want). Joel. On Wed, Sep 26, 2018 at 14:56 Carsten Bock wrote: > Hi, > > I am using FreeSwitch Release: > FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451 > 2018-08-31 19:05:39Z 64bit) is ready > > I have the following, rather simple scenario, where I can't figure > out, how to configure it properly. I use FreeSwitch as Transcoding > B2B-UA/SBC. All the logic is done on a Kamailio in Front of > FreeSwitch. Thus my dialplan is rather simple: > > > > > data="sofia/internal/${destination_number}@${loadbalancer}"/> > > > > > In order to provide transcoding, I have set the following in my > internal.xml profile: > > > > > > > > > > > > > > In vars.xml, I have the following: > > > > > > > > > This works like a charm for all Voice-Calls, the B-Party gets an offer > including all the codecs from the list, e.g. A-Party supports only > G722, B-Party supports only PCMU, so FreeSwitch does the transcoding. > > However, it leads to issues, when it comes to Video. As soon as I add > H264 and VP8 to the codecs list (global_codec_pref / > outbound_codec_prefs), the call to my B-Party includes the offer for > Video as well, even if the A-Party did not offer Video initially. Is > there an easy way to solve this? What are the right settings for such > scenario? > > I tried the following in my dialplan: > > > > data="nolocal:absolute_codec_string=G722,PCMA,PCMU"/> > > > This would remove the Video from the initial invite, if the A-Party > did not offer Video initially. However, if I then enable Video at a > later stage (Re-INVITE with m=video), FreeSwitch will not forward the > new stream to the B-Party and then instead just send a re-INVITE with > Audio only to the B-Party. > > Where did I go wrong? Any ideas? > > Thanks, > Carsten > > -- > > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Millerntorplatz 1 > 20359 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten at ng-voice.com > > Office +49 40 5247593-40 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch Thu Sep 27 12:04:19 2018 From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Tim Jones) Date: Thu, 27 Sep 2018 12:04:19 +0000 Subject: Unable to transfer to automated extension Message-ID: Hi, Normal transfers to other extensions work fine (e.g. from one handset to another). However, we're getting tired of telepests and I've created a special extension. The idea is it saves you getting annoyed and shouting, you just transfer to the pest extension. However when attempting that transfer, all that happens is that it cuts the caller off. It doesn't matter if we try blind or normal transfers. Evidently I'm missing a config line or two ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch Thu Sep 27 12:09:29 2018 From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Tim Jones) Date: Thu, 27 Sep 2018 12:09:29 +0000 Subject: Unable to transfer to automated extension Message-ID: (Apologies if this comes through twice on the list, but I can't see the original in my Inbox from list or my Sent items !) I've got a FS system where normal transfers (i.e. handset to handset) work fine. However, in an effort to combat telepests without raising blood pressure amongst people, I've created a special pest extension. The idea being you just transfer the pests to a pre-recorded announcement. However, all that happens is that you the caller gets cut-off. So I guess I'm missing an important config line or two ? From joel at textplus.com Thu Sep 27 09:17:36 2018 From: joel at textplus.com (Joel Serrano) Date: Thu, 27 Sep 2018 11:17:36 +0200 Subject: [Freeswitch-users] Video Upgrade/Downgrade and Transcoding In-Reply-To: References: Message-ID: Hi Carsten.. I checked and I have the codecs settings like you.. Can't help more there sorry. Joel. On Thu, Sep 27, 2018 at 11:02 AM Joel Serrano wrote: > Hi Carsten, > > I have a setup exactly like that... I’ll get you the combination of > settings I have for FS to handle the codecs. > > As for your second problem, I can tell you it will not work, I have a > quote from FS devs to implement the reinvite with video but as of right now > you cannot switch from audio to audio+video (unless you proxy_media which > of course you don’t want). > > Joel. > > On Wed, Sep 26, 2018 at 14:56 Carsten Bock wrote: > >> Hi, >> >> I am using FreeSwitch Release: >> FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451 >> 2018-08-31 19:05:39Z 64bit) is ready >> >> I have the following, rather simple scenario, where I can't figure >> out, how to configure it properly. I use FreeSwitch as Transcoding >> B2B-UA/SBC. All the logic is done on a Kamailio in Front of >> FreeSwitch. Thus my dialplan is rather simple: >> >> >> >> >> > data="sofia/internal/${destination_number}@${loadbalancer}"/> >> >> >> >> >> In order to provide transcoding, I have set the following in my >> internal.xml profile: >> >> >> >> >> >> >> >> >> >> >> >> >> >> In vars.xml, I have the following: >> >> >> >> >> >> >> >> >> This works like a charm for all Voice-Calls, the B-Party gets an offer >> including all the codecs from the list, e.g. A-Party supports only >> G722, B-Party supports only PCMU, so FreeSwitch does the transcoding. >> >> However, it leads to issues, when it comes to Video. As soon as I add >> H264 and VP8 to the codecs list (global_codec_pref / >> outbound_codec_prefs), the call to my B-Party includes the offer for >> Video as well, even if the A-Party did not offer Video initially. Is >> there an easy way to solve this? What are the right settings for such >> scenario? >> >> I tried the following in my dialplan: >> >> >> >> > data="nolocal:absolute_codec_string=G722,PCMA,PCMU"/> >> >> >> This would remove the Video from the initial invite, if the A-Party >> did not offer Video initially. However, if I then enable Video at a >> later stage (Re-INVITE with m=video), FreeSwitch will not forward the >> new stream to the B-Party and then instead just send a re-INVITE with >> Audio only to the B-Party. >> >> Where did I go wrong? Any ideas? >> >> Thanks, >> Carsten >> >> -- >> >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH >> Millerntorplatz 1 >> 20359 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten at ng-voice.com >> >> Office +49 40 5247593-40 >> Fax +49 40 5247593-99 >> >> Sitz der Gesellschaft: Hamburg >> Registergericht: Amtsgericht Hamburg, HRB 120189 >> Geschäftsführer: Carsten Bock >> Ust-ID: DE279344284 >> >> Hier finden Sie unsere handelsrechtlichen Pflichtangaben: >> http://www.ng-voice.com/imprint/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From modesto916 at gmail.com Thu Sep 27 15:06:28 2018 From: modesto916 at gmail.com (Antonio Modesto) Date: Thu, 27 Sep 2018 12:06:28 -0300 Subject: [Freeswitch-users] RTP problems when using "soft" timer Message-ID: Hi everyone, I have had some problems lately with poor call quality when using FXO channels (mod_khomp) talking to sip phones. Using wireshark I noticed that the RTP stream sent from FS to my SIP device had some packets with wrong timestamps. After changing the rtp timer from soft to posix, the problem was solved. Do you guys know if there is any side effect in using the posix timer? *I am running FS 1.6.20 on a XenServer VM with Debian (Kernel 3.16.0). I can't upgrade to FS 1.8 because Khomp still does not support it. -- Antônio Modesto -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Sep 28 09:49:25 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 28 Sep 2018 09:49:25 +0000 Subject: [Freeswitch-users] Attended Transfer to Call Center Queue when agent answers at the same time the transfer is completed In-Reply-To: References: <9F98F3C8-53E1-476C-A518-4C84490A2ACA@jerris.com> , <06BAF44D-0D70-42DB-884E-BD2AF879C83B@jerris.com>, , Message-ID: We're offering a $100 bounty to fix this if anyone is interested. https://freeswitch.org/jira/browse/FS-11365 Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 29 August 2018 11:01:17 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Attended Transfer to Call Center Queue when agent answers at the same time the transfer is completed After more testing we've identified this was a device specific issue caused by a change we made in mod_sofa.c to update the display on attended transfer to Bria. We had tried the existing methods of updating the display on attended transfer using SIP INFO and SIP UPDATE but these do not work with Bria, the INVITE was the only method we could get to work. It would be great if someone could help fix this, we will raise a JIRA. Thanks, Shaun ________________________________ From: Shaun Stokes Sent: 29 August 2018 08:37:03 To: FreeSWITCH Users Help Subject: Attended Transfer to Call Center Queue when agent answers at the same time the transfer is completed Hi All, FreeSWITCH 1.6.20 When performing an Attended Transfer to a Call Center Queue if the agent answers at the same time the transfer is completed the call is dropped and both the agent and transferee are sent the SIP message Destination_Out_Of_Order. The call disappears from active calls\sessions but still exists as a member in the Call Center Queue, the only way to clear this call is to either manually delete the call from the queue members table in the backend FS database, reload the Call Center module or restart FreeSWITCH. It's very difficult to time correctly, even when attempting to answer and transfer the call at the same time this only occurs in roughly every 1 in 20 calls. However, in call centre environments where calls may be regularly transferred into queues this occurs frequently. Training agents to use the appropriate transfer method is one approach but is not fix. Is this a known problem? Yet to test this on FreeSWITCH 1.8 and master. Thanks, Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Sep 28 16:38:52 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 28 Sep 2018 12:38:52 -0400 Subject: [Freeswitch-users] Encrypted video calls demo In-Reply-To: References: Message-ID: <1285D1BF-CDFC-4921-8E44-8832B9535ECE@jerris.com> 1.8 on debian 9 is released and works well from our repos but I dont think we have install docs updated yet, that is coming very soon. > On Sep 25, 2018, at 12:25 PM, Joel Serrano wrote: > > Hi All, > > I friend recently asked me about a PoC he is setting up and he needs encrypted video calls..blablabla. > > Long story short, I suggested to give FS + Verto a try blablabla.. but I wasn't sure if the "easy getting started" today would be debian8+fs16 or debian9+fs18... > > So my question is, for a really quick demo of FS+Verto would you go with 1.6 and debian8 or 1.8 and debian9 (for a first-time-fs-user)? > > I personally would test with v1.8 but not sure if the installation and can still be a little more tricky than v1.6 on debian8...? > > Opinions?? :D -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Fri Sep 28 16:59:00 2018 From: social at bohboh.info (Social Boh) Date: Fri, 28 Sep 2018 11:59:00 -0500 Subject: [Freeswitch-users] Verto Error Message-ID: <420bc624-a60b-7c67-e5fc-09cf5f45f2db@bohboh.info> Hello list, I can't trace and solve a problem with Verto: the error is: Client Connect from 1.2.3.4:61079 accepted [DEBUG] mod_verto.c:1992 1.2.3.4:61079 Starting client thread. [DEBUG] mod_verto.c:1839 1.2.3.4:61079 *WS SETUP FAILED* [DEBUG] mod_verto.c:2019 1.2.3.4:61079 Ending client thread. [DEBUG] mod_verto.c:2027 1.2.3.4:61079 Thread ended I found similar threads without solution. Let's Encrypt Certificate is for two sub domains and then: *cat fullchain.pem privkey.pem > wss.pem* Verto Configuration standard. *verto debug=10, fsctl loglevel 7, console loglevel 7* but no more messages on FS console. Thank you. Regards -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Fri Sep 28 21:19:31 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Sat, 29 Sep 2018 04:19:31 +0700 Subject: [Freeswitch-users] Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect ) In-Reply-To: References: Message-ID: Dear Brain West, thank you for your response. I would like to confirm that either using export or set on a leg of "rtp_secure_media=true" with the following dial-string is not working for me. One leg call is fine but it does not work for 2-leg call (I could not hear the sound and the call terminates after {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" On Wed, 1 Aug 2018 at 23:20, Brian West wrote: > don't us export, set it inside {}, or on use set on a-leg. > > /b > > > On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra > wrote: > >> Hello, >> >> Currently, I faced a problem regarding SRTP outbound call to user (Leg B). >> >> The scenario is like this, >> >> - We set up our own root CA to an IP address (e.g 192.168.0.13) >> - We create a server certificate for freeswitch at 192.168.0.13 >> - Linphone is used as SIP client and is configured to trust our root >> CA by default. >> - Linphone A is configured to register to Freeswitch vis TLS + SRTP. >> (One leg call to server has both SIPs and SRTP – completely secure) >> - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting >> for Linphone A to call to. >> >> (One leg call to server, e.g. 9196 (echo test), is completely secure with >> SRTP + SIPs) >> >> - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg >> B is not encrypted with SRTP and SIPs at all. This causes *SRTP >> unprotect failed with code 7 (auth check failed)**.* >> >> + Dialplan Configuration >> >> >> >> >> >> The dial-string is > data="user/${dialed_extension}@${domain_name}"/> >> >> + Directory Configruation: >> >> > value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ >> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> >> >> My question is that, is there any configuration left that I have to set >> up in order to let freeswitch initiate an outbound call to Leg B correctly >> with SRTP and SIPs (tls)? >> >> Any help would be really appreciated. >> Thank you so much. >> Best Regard, >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sat Sep 29 10:35:02 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 29 Sep 2018 13:35:02 +0300 Subject: [Freeswitch-users] Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect ) In-Reply-To: References: Message-ID: As i understand you try overwrite transport to user B registration. In many case users is located behind NAT and FS cannot establish TLS connections to B-user. Think in your case need to disable all non TLS sockets and then simple try bridge "user/{user}@{domain}" сб, 29 сент. 2018 г. в 13:20, Chhorm Chhatra : > Dear Brain West, > thank you for your response. > I would like to confirm that either using export or set on a leg of > "rtp_secure_media=true" with the following dial-string is not working for > me. One leg call is fine but it does not work for 2-leg call (I could not > hear the sound and the call terminates after > {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ > ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" > > On Wed, 1 Aug 2018 at 23:20, Brian West wrote: > >> don't us export, set it inside {}, or on use set on a-leg. >> >> /b >> >> >> On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra >> wrote: >> >>> Hello, >>> >>> Currently, I faced a problem regarding SRTP outbound call to user (Leg >>> B). >>> >>> The scenario is like this, >>> >>> - We set up our own root CA to an IP address (e.g 192.168.0.13) >>> - We create a server certificate for freeswitch at 192.168.0.13 >>> - Linphone is used as SIP client and is configured to trust our root >>> CA by default. >>> - Linphone A is configured to register to Freeswitch vis TLS + SRTP. >>> (One leg call to server has both SIPs and SRTP – completely secure) >>> - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting >>> for Linphone A to call to. >>> >>> (One leg call to server, e.g. 9196 (echo test), is completely secure >>> with SRTP + SIPs) >>> >>> - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg >>> B is not encrypted with SRTP and SIPs at all. This causes *SRTP >>> unprotect failed with code 7 (auth check failed)**.* >>> >>> + Dialplan Configuration >>> >>> >>> >>> >>> >>> The dial-string is >> data="user/${dialed_extension}@${domain_name}"/> >>> >>> + Directory Configruation: >>> >>> >> value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ >>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ >>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> >>> >>> My question is that, is there any configuration left that I have to set >>> up in order to let freeswitch initiate an outbound call to Leg B correctly >>> with SRTP and SIPs (tls)? >>> >>> Any help would be really appreciated. >>> Thank you so much. >>> Best Regard, >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 <(918)%20424-9378> >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sat Sep 29 10:37:39 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 29 Sep 2018 13:37:39 +0300 Subject: [Freeswitch-users] Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect ) In-Reply-To: References: Message-ID: Need to revert back "dial-string" in directory config Also important "sips" and "sip" uri different. Please make sure you not use sips uri in client side. Sergey сб, 29 сент. 2018 г. в 13:35, Sergey Safarov : > As i understand you try overwrite transport to user B registration. > In many case users is located behind NAT and FS cannot establish TLS > connections to B-user. > > Think in your case need to disable all non TLS sockets and then simple try > bridge "user/{user}@{domain}" > > сб, 29 сент. 2018 г. в 13:20, Chhorm Chhatra : > >> Dear Brain West, >> thank you for your response. >> I would like to confirm that either using export or set on a leg of >> "rtp_secure_media=true" with the following dial-string is not working for >> me. One leg call is fine but it does not work for 2-leg call (I could not >> hear the sound and the call terminates after >> {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ >> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" >> >> On Wed, 1 Aug 2018 at 23:20, Brian West wrote: >> >>> don't us export, set it inside {}, or on use set on a-leg. >>> >>> /b >>> >>> >>> On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra >>> wrote: >>> >>>> Hello, >>>> >>>> Currently, I faced a problem regarding SRTP outbound call to user (Leg >>>> B). >>>> >>>> The scenario is like this, >>>> >>>> - We set up our own root CA to an IP address (e.g 192.168.0.13) >>>> - We create a server certificate for freeswitch at 192.168.0.13 >>>> - Linphone is used as SIP client and is configured to trust our >>>> root CA by default. >>>> - Linphone A is configured to register to Freeswitch vis TLS + >>>> SRTP. (One leg call to server has both SIPs and SRTP – completely secure) >>>> - Linphone B is registered to Freeswitch via TLS + SRTP, and >>>> waiting for Linphone A to call to. >>>> >>>> (One leg call to server, e.g. 9196 (echo test), is completely secure >>>> with SRTP + SIPs) >>>> >>>> - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but >>>> Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP >>>> unprotect failed with code 7 (auth check failed)**.* >>>> >>>> + Dialplan Configuration >>>> >>>> >>>> >>>> >>>> >>>> The dial-string is >>> data="user/${dialed_extension}@${domain_name}"/> >>>> >>>> + Directory Configruation: >>>> >>>> >>> value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ >>>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ >>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> >>>> >>>> My question is that, is there any configuration left that I have to set >>>> up in order to let freeswitch initiate an outbound call to Leg B correctly >>>> with SRTP and SIPs (tls)? >>>> >>>> Any help would be really appreciated. >>>> Thank you so much. >>>> Best Regard, >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>>> >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> >>> Need Commercial support? email sales at freeswitch.com >>> >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> >>> >>> Email: brian at freeswitch.com >>> >>> Mobile: 918-424-9378 <(918)%20424-9378> >>> >>> Website: https://www.FreeSWITCH.com >>> >>> [image: https://www.facebook.com/signalwireinc?src=email] >>> [image: >>> https://twitter.com/freeswitch] >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Sat Sep 29 11:02:12 2018 From: joel at textplus.com (Joel Serrano) Date: Sat, 29 Sep 2018 13:02:12 +0200 Subject: [Freeswitch-users] Encrypted video calls demo In-Reply-To: <1285D1BF-CDFC-4921-8E44-8832B9535ECE@jerris.com> References: <1285D1BF-CDFC-4921-8E44-8832B9535ECE@jerris.com> Message-ID: Awesome!! Thanks Mike!! On Sat, Sep 29, 2018 at 12:02 Michael Jerris wrote: > 1.8 on debian 9 is released and works well from our repos but I dont think > we have install docs updated yet, that is coming very soon. > > On Sep 25, 2018, at 12:25 PM, Joel Serrano wrote: > > Hi All, > > I friend recently asked me about a PoC he is setting up and he needs > encrypted video calls..blablabla. > > Long story short, I suggested to give FS + Verto a try blablabla.. but I > wasn't sure if the "easy getting started" *today* would be debian8+fs16 > or debian9+fs18... > > So my question is, for a really quick demo of FS+Verto would you go with > 1.6 and debian8 or 1.8 and debian9 (for a first-time-fs-user)? > > I personally would test with v1.8 but not sure if the installation and can > still be a little more tricky than v1.6 on debian8...? > > Opinions?? :D > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Sat Sep 29 12:09:44 2018 From: social at bohboh.info (Social Boh) Date: Sat, 29 Sep 2018 07:09:44 -0500 Subject: [Freeswitch-users] Verto Error In-Reply-To: <420bc624-a60b-7c67-e5fc-09cf5f45f2db@bohboh.info> References: <420bc624-a60b-7c67-e5fc-09cf5f45f2db@bohboh.info> Message-ID: <305967cb-e5af-1c7e-9d9e-91d30df6ad9d@bohboh.info> My mistake. Certificate error. Sorry for the noise. Regards --- I'm SoCIaL, MayBe El 28/09/2018 a las 11:59, Social Boh escribió: > > Hello list, > > I can't trace and solve a problem with Verto: the error is: > > Client Connect from 1.2.3.4:61079 accepted > [DEBUG] mod_verto.c:1992 1.2.3.4:61079 Starting client thread. > [DEBUG] mod_verto.c:1839 1.2.3.4:61079 *WS SETUP FAILED* > [DEBUG] mod_verto.c:2019 1.2.3.4:61079 Ending client thread. > [DEBUG] mod_verto.c:2027 1.2.3.4:61079 Thread ended > > I found similar threads without solution. > > Let's Encrypt Certificate is for two sub domains and then: > > *cat fullchain.pem privkey.pem > wss.pem* > > Verto Configuration standard. > > *verto debug=10, fsctl loglevel 7, console loglevel 7* > > but no more messages on FS console. > > Thank you. > > Regards > > > > > -- > --- > I'm SoCIaL, MayBe > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... 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