From bote_radio at botecomm.com Mon Feb 1 00:11:12 2016 From: bote_radio at botecomm.com (Bote Man) Date: Sun, 31 Jan 2016 16:11:12 -0500 Subject: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7] In-Reply-To: References: Message-ID: <00f801d15c6b$ea5bdd00$bf139700$@botecomm.com> Pastebin is preferred to posting miles of log files. https://pastebin.freeswitch.org/ Select ?FreeSWITCH log? format for easier troubleshooting. Thanks. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: SamyGo Sent: Sunday, 31 January, 2016 14:48 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7] Hi Bilal, I have FS 1.7 and I get video calls going on it without trouble. Could you post the CLI logs as well as the SIP trace appearing for the all..only the First INVITE from caller and 200OK should suffice. Can you tell if you either mod_h26x or mod_av loaded in FS. Also load mod_vpx and then try making calls. FS 1.6 is now part of newer versions so dont need to explicitly install 1m6 for video only. Regards, Sammy On Jan 30, 2016 12:09, "Bilal Abbasi" wrote: Hi Users, I am writing this question after reading old questions and forums, so my problem is i am unable to dial video call using freeswitch 1.7. I have enabled video codecs in var.xml (VP8,H263,H264) and i am using bria sipfone for UA1 and UA2. I am able to dial the normal call but unable to dial video. do i need any further variables to activate video calling? One of my friend told me to shift to FS 1.6 as that provides transcoding. so can anybody help me out. FS Version: 1.7.0 OS: jessie 8 Softphone: Bria Regards Abbasi _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160131/3df97a58/attachment.html From omortimer at gmail.com Mon Feb 1 00:58:22 2016 From: omortimer at gmail.com (Oz Mortimer) Date: Sun, 31 Jan 2016 21:58:22 +0000 Subject: [Freeswitch-users] extra header account code is not written to cdr if cancel is received a few ms after invite In-Reply-To: <3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com> References: <56AE42E1.3080703@gmail.com> <3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com> Message-ID: <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com> Try export rather than set > On 31 Jan 2016, at 18:45, servtelar at gmail.com wrote: > > Shouldn't that be done as inline? > > Sent from my iPhone > >> On Jan 31, 2016, at 12:22 PM, Panagiotis Skoulikaritis wrote: >> >> Dear all >> >> I have an implementation FreeSWITCH as a sort of SBC, it is used to send >> the calls to the terminating carriers and do topology hiding, nothing >> fancy. Also I gather cdrs from the FreeSWITCH. >> >> In order to distinguish each customer on the FS cdrs I send an extra >> header containing the accountcode. >> >> I have noticed that if the call is canceled immediately on the same sec, >> the account code is not written on the cdr. >> To be more precise the cancel is send a few milliseconds after it has >> received the invite, and before the FreeSWITCH has sent the call to the >> terminating carrier (I'm using Homer Sipcapture to capture all the >> traces and I don't see an attempt being made at the terminating carrier) >> also I don't see a b-leg cdr. >> >> FreeSWITCH is writing both a-leg and b-leg cdrs in csv format. >> >> The dialplan that I use is simple >> >> >> > expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> any idea how I can make sure that the account code will always be written ? >> >> >> Best Regards >> >> Panagiotis >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bilaln018 at gmail.com Mon Feb 1 08:08:46 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Mon, 1 Feb 2016 10:08:46 +0500 Subject: [Freeswitch-users] [Video Calling Issue][Freeswitch 1.7] In-Reply-To: References: Message-ID: Hi sammy, Thanks for your reply, yes i have enabled mod_h26x and mod_vpx loaded in modules.conf as well.i have enabled mod_av as well. ok i will share the logs and dump file. Regards Abbasi On Mon, Feb 1, 2016 at 12:47 AM, SamyGo wrote: > Hi Bilal, > I have FS 1.7 and I get video calls going on it without trouble. Could you > post the CLI logs as well as the SIP trace appearing for the all..only the > First INVITE from caller and 200OK should suffice. > Can you tell if you either mod_h26x or mod_av loaded in FS. Also load > mod_vpx and then try making calls. > FS 1.6 is now part of newer versions so dont need to explicitly install > 1m6 for video only. > > Regards, > Sammy > On Jan 30, 2016 12:09, "Bilal Abbasi" wrote: > >> Hi Users, >> >> I am writing this question after reading old questions and forums, so my >> problem is i am unable to dial video call using freeswitch 1.7. >> I have enabled video codecs in var.xml (VP8,H263,H264) and i am using >> bria sipfone for UA1 and UA2. >> I am able to dial the normal call but unable to dial video. >> do i need any further variables to activate video calling? >> One of my friend told me to shift to FS 1.6 as that provides transcoding. >> so can anybody help me out. >> >> FS Version: 1.7.0 >> OS: jessie 8 >> Softphone: Bria >> >> Regards >> Abbasi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/a60c94d4/attachment.html From luis.azedo at factorlusitano.com Mon Feb 1 16:47:22 2016 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Mon, 1 Feb 2016 13:47:22 +0000 Subject: [Freeswitch-users] recording problems with mod_shout debian/master Message-ID: Hi, anyone having problems recording in mp3 format ? 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD <96722133-5060-508 at BJC.BGI.CG.BD> start /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)* 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match requested rate 16000* 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com <995582142 at teste.sip.90e9.com>* 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence detection configured; assuming start of speech* 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits (http://lame.sf.net )* 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass filter disabled* 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/1af0633f/attachment.html From tg at level5.de Mon Feb 1 17:12:25 2016 From: tg at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Mon, 1 Feb 2016 15:12:25 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js Message-ID: <56AF67C9.2010700@level5.de> Hi, I want to setup a Click-2-Call-Button for our website. Is there any significant difference between Verto-Mod and libraries such as SIP.js? I do not really understand the advantage of Verto (if there is any). Thanks in advance, Thorsten From yadenis at seznam.cz Mon Feb 1 17:25:39 2016 From: yadenis at seznam.cz (Denis Jakovlev) Date: Mon, 1 Feb 2016 15:25:39 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <56AF67C9.2010700@level5.de> References: <56AF67C9.2010700@level5.de> Message-ID: <95599933.20160201152539@seznam.cz> Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/ec538112/attachment.html From DEdwards at vertical.com Mon Feb 1 17:34:42 2016 From: DEdwards at vertical.com (Dan Edwards) Date: Mon, 1 Feb 2016 14:34:42 +0000 Subject: [Freeswitch-users] WebSocket behind NGINX In-Reply-To: <56AD0CE7.6000607@gmail.com> References: <56AD0CE7.6000607@gmail.com> Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997A4E4@PHXEX2.vertical.com> I'm also running behind Nginx and what I found worked was to proxy to the actual IP address (192.168.1.1 vs. 127.0.0.1), then explicitly removing 192.168.1.1 from the localnet ACL in acl.conf. I had to remove 192.168.1.1 from localnet so FS will offer external IP addresses for RTP. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton Sent: Saturday, January 30, 2016 2:20 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] WebSocket behind NGINX Hello All, I have to proxy all websocket requests though a nginx server. Right now I am using next configuration: map $http_upgrade $connection_upgrade { default upgrade; '' close; } server { listen 443; server_name wss.somedomain.com.ua; ssl on; ssl_certificate /etc/nginx/cert.pem; ssl_certificate_key /etc/nginx/private.key; location / { proxy_pass http://127.0.0.1:5066; proxy_http_version 1.1; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; proxy_read_timeout 86400s; } access_log /var/log/nginx/wss_access; error_log /var/log/nginx/wss_error debug; } I dumped traffic from nginx and found out that "switching protocol" phrase was successful but INVITE message from my browser in pending state. Maybe FreeSWITCH wants real IP not loopback? Who have faced with similar problem? BR, Anton _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bote_radio at botecomm.com Mon Feb 1 17:42:48 2016 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 1 Feb 2016 09:42:48 -0500 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <95599933.20160201152539@seznam.cz> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> Message-ID: <003a01d15cfe$d26c3350$774499f0$@botecomm.com> Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/261f1571/attachment-0001.html From eschmidbauer at gmail.com Mon Feb 1 18:16:45 2016 From: eschmidbauer at gmail.com (E. Schmidbauer) Date: Mon, 1 Feb 2016 10:16:45 -0500 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> Message-ID: Can anyone share the VM memory/cpu specs used in these cases? We want to run around 300 CPS on FS (running on vmware). very little transcoding (if any), audio only How much memory/cpu should be provisioned? I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much memory? Thanks, E On Fri, Jan 29, 2016 at 10:00 PM, servtelar wrote: > Thanks a lot guys for sharing this info. It?s really helpful. > > > On Jan 28, 2016, at 6:18 PM, Sergey Safarov wrote: > > We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load. > > Sergey > > On Fri, Jan 29, 2016 at 2:01 AM, servtelar wrote: > >> Hi Chad >> >> How many legs you are handling with 20 cores on a conference? >> >> Regards >> >> Gustavo >> >> On Jan 28, 2016, at 7:55 PM, Chad Phillips >> wrote: >> >> I've had very good luck running the newer video branch code on >> ProfitBricks: https://www.profitbricks.com/ >> >> As far as I understand, the CPU cycles are guaranteed on their platform. >> I've had to put as many as 20 cores on a server to handle some of our >> busier video conference calls, but with that it runs quite smoothly. >> >> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards >> wrote: >> >>> I am reviewing the Confluence Virtualization page and had some >>> questions, in particular about VMWare. My company distributes some of its >>> software as a VMWare image file and we were looking to distribute a new >>> product using FS in the same manner. The products operate at a customer >>> premise, on their VMWare infrastructure, not in a cloud environment. Since >>> our customers already have VMWare, switching to a different VM >>> infrastructure is going to hurt, so I am looking for options/alternatives. >>> >>> First, does anybody know if the virtual timing issues with VMWare have >>> improved since this page was last updated in 2014? Is VMWare still not good >>> enough? Is it possible to throw CPU & memory at this and make VMWare good >>> enough, or is the virtual timing just not workable? >>> >>> On the virtualization page, there was a comment from 2010 that you might >>> be happy with a High CPU Medium instance on AWS EC2. Certainly workload is >>> a factor here, but I am trying to get my head around how big a machine to >>> perform how small a workload. Is there a place where people talk about >>> their experiences? >>> >>> Are there other VM platforms that might be acceptable? >>> >>> Any help or comment is appreciated. >>> >>> Thank you, >>> Dan >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/5879268d/attachment.html From llribeiro90 at gmail.com Mon Feb 1 18:38:25 2016 From: llribeiro90 at gmail.com (Leonardo Lima Ribeiro) Date: Mon, 1 Feb 2016 13:38:25 -0200 Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing In-Reply-To: References: Message-ID: Thanks Chad and Oz. I tried two things: 1) Oz suggestion: local new_session = freeswitch.Session("[origination_caller_id_name='7136694967',origination_caller_id_number='7136694967']sofia/gateway/4_NEXTIVA/3157244022", session); new_session:setVariable(?RECORD_STEREO?,?false?) new_session:sleep("5000") new_session:streamFile("voicemail/vm-goodbye.wav") new_session:sleep("10000") new_session:hangup() Did not work? 2) Chad suggestion: local new_session = freeswitch.Session("[origination_caller_id_name='7136694967',origination_caller_id_number='7136694967']sofia/gateway/4_NEXTIVA/3157244022", session); uuid = new_session:get_uuid() api = freeswitch.API() reply = api:executeString("uuid_record " .. uuid .. " start /usr/local/freeswitch/recordings/myrecording.wav") new_session:sleep("5000") new_session:streamFile("voicemail/vm-goodbye.wav") new_session:sleep("10000") new_session:hangup() Did not work too? Both cases I have 5 seconds of silence (first 5000 pause) and then when I play the Good Bye wav file it starts to record normally. So I have 5 seconds of silence, good bye sound and 10 seconds of recording. Chad, I can?t use session:recordFile because it?s a synchronous command that blocks my script.. I need to do a lot of actions like pauses, stream files etc., while the call is still going on? Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if it exists hehe, maybe a github issue?) Thank you, > On Jan 30, 2016, at 12:36 AM, Chad Phillips wrote: > > have you tried the session method specifically for recording files? > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile > > or if that doesn't work, maybe this: > > local new_session = freeswitch.Session("someoriginatestring", session); > uuid = new_session:get_uuid() > api = freeswitch.API() > reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath) > > On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer > wrote: > since this is a single left call try setting record_stereo to false - https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO > I'm probably wrong, but worth a shot! > > On 29 Jan 2016, at 19:53, Leonardo Ribeiro > wrote: > >> Hello Guys, >> >> Any idea? >> I could not evolve this yet... >> >> Thank you, >> >> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro >: >> Hello all, >> >> I?m trying to record an IVR using my gateway to do the outbound call in my luascript: >> >> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 ", session); >> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav") >> new_session:sleep("10000") >> new_session:hangup() >> >> So in the above script I just call to the Bank Of America as an example and try to record the first 10 seconds of the call in the recordings path. >> >> The problem is that I have an empty recording file.. Why? >> >> The funny thing is: if I add this command after the record_session command: >> new_session:streamFile("voicemail/vm-goodbye.wav?); >> >> And then this is the entire new script: >> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 ", session); >> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?) >> new_session:streamFile("voicemail/vm-goodbye.wav?); >> new_session:sleep("10000") >> new_session:hangup() >> >> I can hear the Good Bye sound from my script and then hear the Bank of America IVR. >> >> I just don?t understand why the record works if I play a sound in our side and the record does not work if I don?t play any sound. >> >> Do you know what?s happening? How can I solve this? >> >> Thank you! >> >> >> >> >> -- >> Leonardo Lima Ribeiro >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/a1a8bfd1/attachment-0001.html From yadenis at seznam.cz Mon Feb 1 18:30:13 2016 From: yadenis at seznam.cz (Denis Jakovlev) Date: Mon, 1 Feb 2016 16:30:13 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <003a01d15cfe$d26c3350$774499f0$@botecomm.com> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> Message-ID: <1105687706.20160201163013@seznam.cz> Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/5b6b3a3b/attachment.html From lexxua at gmail.com Mon Feb 1 18:34:06 2016 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Mon, 1 Feb 2016 16:34:06 +0100 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> Message-ID: Hello, it depends. In my case 24 threads and 8gb of ram was quite enough. But I used only xml-dialplan and hash counters. On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer wrote: > Can anyone share the VM memory/cpu specs used in these cases? > We want to run around 300 CPS on FS (running on vmware). > very little transcoding (if any), audio only > How much memory/cpu should be provisioned? > I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much > memory? > Thanks, > E > > On Fri, Jan 29, 2016 at 10:00 PM, servtelar wrote: > >> Thanks a lot guys for sharing this info. It?s really helpful. >> >> >> On Jan 28, 2016, at 6:18 PM, Sergey Safarov wrote: >> >> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load. >> >> Sergey >> >> On Fri, Jan 29, 2016 at 2:01 AM, servtelar wrote: >> >>> Hi Chad >>> >>> How many legs you are handling with 20 cores on a conference? >>> >>> Regards >>> >>> Gustavo >>> >>> On Jan 28, 2016, at 7:55 PM, Chad Phillips >>> wrote: >>> >>> I've had very good luck running the newer video branch code on >>> ProfitBricks: https://www.profitbricks.com/ >>> >>> As far as I understand, the CPU cycles are guaranteed on their platform. >>> I've had to put as many as 20 cores on a server to handle some of our >>> busier video conference calls, but with that it runs quite smoothly. >>> >>> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards >>> wrote: >>> >>>> I am reviewing the Confluence Virtualization page and had some >>>> questions, in particular about VMWare. My company distributes some of its >>>> software as a VMWare image file and we were looking to distribute a new >>>> product using FS in the same manner. The products operate at a customer >>>> premise, on their VMWare infrastructure, not in a cloud environment. Since >>>> our customers already have VMWare, switching to a different VM >>>> infrastructure is going to hurt, so I am looking for options/alternatives. >>>> >>>> First, does anybody know if the virtual timing issues with VMWare have >>>> improved since this page was last updated in 2014? Is VMWare still not good >>>> enough? Is it possible to throw CPU & memory at this and make VMWare good >>>> enough, or is the virtual timing just not workable? >>>> >>>> On the virtualization page, there was a comment from 2010 that you >>>> might be happy with a High CPU Medium instance on AWS EC2. Certainly >>>> workload is a factor here, but I am trying to get my head around how big a >>>> machine to perform how small a workload. Is there a place where people talk >>>> about their experiences? >>>> >>>> Are there other VM platforms that might be acceptable? >>>> >>>> Any help or comment is appreciated. >>>> >>>> Thank you, >>>> Dan >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/28be4cfb/attachment.html From eschmidbauer at gmail.com Mon Feb 1 18:38:29 2016 From: eschmidbauer at gmail.com (E. Schmidbauer) Date: Mon, 1 Feb 2016 10:38:29 -0500 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> Message-ID: Hi Volodymyr, We are using xml-curl (i dont think that should have too much affect) and hash counters only. No database connections, etc... Thanks-- 24 threads and 8gb seems like a good start! Emmanuel On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov wrote: > Hello, it depends. > In my case 24 threads and 8gb of ram was quite enough. But I used only > xml-dialplan and hash counters. > > On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer > wrote: > >> Can anyone share the VM memory/cpu specs used in these cases? >> We want to run around 300 CPS on FS (running on vmware). >> very little transcoding (if any), audio only >> How much memory/cpu should be provisioned? >> I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how >> much memory? >> Thanks, >> E >> >> On Fri, Jan 29, 2016 at 10:00 PM, servtelar wrote: >> >>> Thanks a lot guys for sharing this info. It?s really helpful. >>> >>> >>> On Jan 28, 2016, at 6:18 PM, Sergey Safarov wrote: >>> >>> We have to core ESXi vm with 140 session (70 calls) with have 70 CPU >>> load. >>> >>> Sergey >>> >>> On Fri, Jan 29, 2016 at 2:01 AM, servtelar wrote: >>> >>>> Hi Chad >>>> >>>> How many legs you are handling with 20 cores on a conference? >>>> >>>> Regards >>>> >>>> Gustavo >>>> >>>> On Jan 28, 2016, at 7:55 PM, Chad Phillips >>>> wrote: >>>> >>>> I've had very good luck running the newer video branch code on >>>> ProfitBricks: https://www.profitbricks.com/ >>>> >>>> As far as I understand, the CPU cycles are guaranteed on their >>>> platform. I've had to put as many as 20 cores on a server to handle some of >>>> our busier video conference calls, but with that it runs quite smoothly. >>>> >>>> On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards >>>> wrote: >>>> >>>>> I am reviewing the Confluence Virtualization page and had some >>>>> questions, in particular about VMWare. My company distributes some of its >>>>> software as a VMWare image file and we were looking to distribute a new >>>>> product using FS in the same manner. The products operate at a customer >>>>> premise, on their VMWare infrastructure, not in a cloud environment. Since >>>>> our customers already have VMWare, switching to a different VM >>>>> infrastructure is going to hurt, so I am looking for options/alternatives. >>>>> >>>>> First, does anybody know if the virtual timing issues with VMWare have >>>>> improved since this page was last updated in 2014? Is VMWare still not good >>>>> enough? Is it possible to throw CPU & memory at this and make VMWare good >>>>> enough, or is the virtual timing just not workable? >>>>> >>>>> On the virtualization page, there was a comment from 2010 that you >>>>> might be happy with a High CPU Medium instance on AWS EC2. Certainly >>>>> workload is a factor here, but I am trying to get my head around how big a >>>>> machine to perform how small a workload. Is there a place where people talk >>>>> about their experiences? >>>>> >>>>> Are there other VM platforms that might be acceptable? >>>>> >>>>> Any help or comment is appreciated. >>>>> >>>>> Thank you, >>>>> Dan >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Best regards, > Volodymyr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/e759166a/attachment-0001.html From brian at freeswitch.org Mon Feb 1 18:56:20 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Feb 2016 09:56:20 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <1105687706.20160201163013@seznam.cz> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> Message-ID: Thats a knee slapper, When did open source for you to modify considered vendor lock-in? On Mon, Feb 1, 2016 at 9:30 AM, Denis Jakovlev wrote: > Dobr? den, > > i think this Verto its too much VendorLock :) > > > > > > > > > *-- S pozdravem, Ing.Denis Jakovlev mob.tel > . 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: > * > > Denis, what difficulties did you experience with Verto? > > Thanks. > > Bote > > > *From:* Denis Jakovlev > *Sent:* Monday, 01 February, 2016 09:26 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > I use sip.js. It turned out to be a lot easier that Verto. > > > > > > > > > > > > > > > > > > > > > *-- S pozdravem, Ing.Denis Jakovlev mob.tel > . 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > > significant difference between Verto-Mod and libraries such as SIP.js? > > I do not really understand the advantage of Verto (if there is any). > > Thanks in advance, > Thorsten > > _________________________________________________________________________ * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/599dd79e/attachment.html From abaci64 at gmail.com Mon Feb 1 18:57:34 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 1 Feb 2016 10:57:34 -0500 Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing In-Reply-To: References: Message-ID: did you try to answer the new_session before starting the record? On Mon, Feb 1, 2016 at 10:38 AM, Leonardo Lima Ribeiro < llribeiro90 at gmail.com> wrote: > Thanks Chad and Oz. > > I tried two things: > > 1) Oz suggestion: > > local new_session = freeswitch.Session("[origination_caller_id_name=' > 7136694967',origination_caller_id_number='7136694967 > ']sofia/gateway/4_NEXTIVA/3157244022", session); > new_session:setVariable(?RECORD_STEREO?,?false?) > new_session:sleep("5000") > new_session:streamFile("voicemail/vm-goodbye.wav") > new_session:sleep("10000") > new_session:hangup() > > Did not work? > > 2) Chad suggestion: > > local new_session = freeswitch.Session("[origination_caller_id_name=' > 7136694967',origination_caller_id_number='7136694967 > ']sofia/gateway/4_NEXTIVA/3157244022", session); > > uuid = new_session:get_uuid() > api = freeswitch.API() > reply = api:executeString("uuid_record " .. uuid .. " start > /usr/local/freeswitch/recordings/myrecording.wav") > > new_session:sleep("5000") > new_session:streamFile("voicemail/vm-goodbye.wav") > new_session:sleep("10000") > new_session:hangup() > > Did not work too? > > Both cases I have 5 seconds of silence (first 5000 pause) and then when I > play the Good Bye wav file it starts to record normally. So I have 5 > seconds of silence, good bye sound and 10 seconds of recording. > > Chad, I can?t use session:recordFile because it?s a synchronous command > that blocks my script.. I need to do a lot of actions like pauses, stream > files etc., while the call is still going on? > > Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if > it exists hehe, maybe a github issue?) > > Thank you, > > > On Jan 30, 2016, at 12:36 AM, Chad Phillips > wrote: > > have you tried the session method specifically for recording files? > > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile > > or if that doesn't work, maybe this: > > local new_session = freeswitch.Session("someoriginatestring", session); > uuid = new_session:get_uuid() > api = freeswitch.API() > reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath) > > On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer wrote: > >> since this is a single left call try setting record_stereo to false - >> https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO >> I'm probably wrong, but worth a shot! >> >> On 29 Jan 2016, at 19:53, Leonardo Ribeiro wrote: >> >> Hello Guys, >> >> Any idea? >> I could not evolve this yet... >> >> Thank you, >> >> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro : >> >>> Hello all, >>> >>> I?m trying to record an IVR using my gateway to do the outbound call in >>> my luascript: >>> >>> local new_session = >>> freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/ >>> 3157244022", session); >>> >>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav") >>> new_session:sleep("10000") >>> new_session:hangup() >>> >>> So in the above script I just call to the Bank Of America as an example >>> and try to record the first 10 seconds of the call in the recordings path. >>> >>> The problem is that I have an empty recording file.. Why? >>> >>> The funny thing is: if I add this command after the record_session >>> command: >>> new_session:streamFile("voicemail/vm-goodbye.wav?); >>> >>> And then this is the entire new script: >>> local new_session = >>> freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/ >>> 3157244022", session); >>> >>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?) >>> new_session:streamFile("voicemail/vm-goodbye.wav?); >>> new_session:sleep("10000") >>> new_session:hangup() >>> >>> I can hear the Good Bye sound from my script and then hear the Bank of >>> America IVR. >>> >>> I just don?t understand why the record works if I play a sound in our >>> side and the record does not work if I don?t play any sound. >>> >>> Do you know what?s happening? How can I solve this? >>> >>> Thank you! >>> >>> >> >> >> -- >> Leonardo Lima Ribeiro >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/0615edfc/attachment-0001.html From krice at freeswitch.org Mon Feb 1 19:11:48 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 1 Feb 2016 10:11:48 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <56AF67C9.2010700@level5.de> References: <56AF67C9.2010700@level5.de> Message-ID: <504f01d15d0b$4199d8c0$c4cd8a40$@freeswitch.org> The difference is, Sip.js is a full sip stack written in JS, Verto is a lot smaller simpler stack to use. If you don?t need SIP (and really most people really don?t) when you have WebRTC. Verto greatly simplifies the calling. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thorsten G?llner Sent: Monday, February 1, 2016 8:12 AM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] Verto vs. SIP.js Hi, I want to setup a Click-2-Call-Button for our website. Is there any significant difference between Verto-Mod and libraries such as SIP.js? I do not really understand the advantage of Verto (if there is any). Thanks in advance, Thorsten _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Mon Feb 1 19:12:43 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 1 Feb 2016 10:12:43 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <1105687706.20160201163013@seznam.cz> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> Message-ID: <505001d15d0b$623021c0$26906540$@freeswitch.org> Really? That's why the whole protocol is OpenSource. not much vendor lock in there. it can be used anywhere. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/ffca31d3/attachment.html From giacomo.vacca at gmail.com Mon Feb 1 19:22:24 2016 From: giacomo.vacca at gmail.com (Giacomo Vacca) Date: Mon, 1 Feb 2016 17:22:24 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <505001d15d0b$623021c0$26906540$@freeswitch.org> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> Message-ID: With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. On 1 February 2016 at 17:12, Ken Rice wrote: > Really? That?s why the whole protocol is OpenSource? not much vendor lock > in there? it can be used anywhere? > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis > Jakovlev > *Sent:* Monday, February 1, 2016 9:30 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js > > > > Dobr? den, > > i think this Verto its too much VendorLock :) > > > > > > > *-- S pozdravem,Ing.Denis Jakovlev mob.tel > . 775-415-382pond?l? 1. ?nora 2016, 15:42:48, napsal jste:* > > Denis, what difficulties did you experience with Verto? > > Thanks. > > Bote > > > *From:* Denis Jakovlev > *Sent:* Monday, 01 February, 2016 09:26 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > I use sip.js. It turned out to be a lot easier that Verto. > > > > > > > > > > > > > > > > > > > > *-- S pozdravem,Ing.Denis Jakovlev mob.tel > . 775-415-382pond?l? 1. ?nora 2016, 15:12:25, napsal jste:> > Hi,> I want to setup a Click-2-Call-Button for our website. Is there any > > significant difference between Verto-Mod and libraries such as SIP.js?> I > do not really understand the advantage of Verto (if there is any).> Thanks > in advance,> Thorsten> > _________________________________________________________________________* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/d44c69ce/attachment.html From bote_radio at botecomm.com Mon Feb 1 19:26:28 2016 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 1 Feb 2016 11:26:28 -0500 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <505001d15d0b$623021c0$26906540$@freeswitch.org> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> Message-ID: <007201d15d0d$4d80cde0$e88269a0$@botecomm.com> Perhaps Denis should see what can be done with Verto on the weekly conference call? https://cantina.freeswitch.org/vc Enjoy. Bote From: Ken Rice Sent: Monday, 01 February, 2016 11:13 Subject: Re: [Freeswitch-users] Verto vs. SIP.js Really? That's why the whole protocol is OpenSource. not much vendor lock in there. it can be used anywhere. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/0085e0fd/attachment-0001.html From bote_radio at botecomm.com Mon Feb 1 19:26:28 2016 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 1 Feb 2016 11:26:28 -0500 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> Message-ID: <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> It seems to me that the XML dialplan routing process would consume more cpu than a short and simple curl request to a database server, no? Anyway, there are probably many things that can be done to streamline performance for a particular application. I know on Windoze it was once suggested to me a long time ago to disable the operating system updates of file modification time stamps, for example. That can add up when a busy system is writing to log files and creating individual CDR files to be read and deleted by an accounting process. I don?t know how this would apply to linux, but it demonstrates how far out of the box one can look for performance improvements. In any case, when discussing virtualization performance it is essential to provide specifics of the instance that runs FreeSWITCH. If someone reports that FS ran very poorly, but does not say that it was a tiny instance that was starved for resources then we can?t evaluate that report fairly. So the specifics in the report by Volodymyr are useful. Please keep them coming! I am compiling these anecdotes on Confluence for others to read in the future. Thanks. Bote From: E. Schmidbauer Sent: Monday, 01 February, 2016 10:38 Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments Hi Volodymyr, We are using xml-curl (i dont think that should have too much affect) and hash counters only. No database connections, etc... Thanks-- 24 threads and 8gb seems like a good start! Emmanuel On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov wrote: Hello, it depends. In my case 24 threads and 8gb of ram was quite enough. But I used only xml-dialplan and hash counters. On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer wrote: Can anyone share the VM memory/cpu specs used in these cases? We want to run around 300 CPS on FS (running on vmware). very little transcoding (if any), audio only How much memory/cpu should be provisioned? I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much memory? Thanks, E On Fri, Jan 29, 2016 at 10:00 PM, servtelar wrote: Thanks a lot guys for sharing this info. It?s really helpful. On Jan 28, 2016, at 6:18 PM, Sergey Safarov wrote: We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load. Sergey On Fri, Jan 29, 2016 at 2:01 AM, servtelar wrote: Hi Chad How many legs you are handling with 20 cores on a conference? Regards Gustavo On Jan 28, 2016, at 7:55 PM, Chad Phillips wrote: I've had very good luck running the newer video branch code on ProfitBricks: https://www.profitbricks.com/ As far as I understand, the CPU cycles are guaranteed on their platform. I've had to put as many as 20 cores on a server to handle some of our busier video conference calls, but with that it runs quite smoothly. On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards wrote: I am reviewing the Confluence Virtualization page and had some questions, in particular about VMWare. My company distributes some of its software as a VMWare image file and we were looking to distribute a new product using FS in the same manner. The products operate at a customer premise, on their VMWare infrastructure, not in a cloud environment. Since our customers already have VMWare, switching to a different VM infrastructure is going to hurt, so I am looking for options/alternatives. First, does anybody know if the virtual timing issues with VMWare have improved since this page was last updated in 2014? Is VMWare still not good enough? Is it possible to throw CPU & memory at this and make VMWare good enough, or is the virtual timing just not workable? On the virtualization page, there was a comment from 2010 that you might be happy with a High CPU Medium instance on AWS EC2. Certainly workload is a factor here, but I am trying to get my head around how big a machine to perform how small a workload. Is there a place where people talk about their experiences? Are there other VM platforms that might be acceptable? Any help or comment is appreciated. Thank you, Dan _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/56db9f05/attachment.html From llribeiro90 at gmail.com Mon Feb 1 19:50:16 2016 From: llribeiro90 at gmail.com (Leonardo Lima Ribeiro) Date: Mon, 1 Feb 2016 14:50:16 -0200 Subject: [Freeswitch-users] Simple LuaScript to Record Session Failing In-Reply-To: References: Message-ID: Answer in what extension? I?m just calling to an outbound IVRs and doing some stuffs like sending dtmfs, playing sounds etc. to interact with them. I?m not using any extensions for this type of calls? Is this really necessary? Thank you, > On Feb 1, 2016, at 1:57 PM, Abaci B wrote: > > did you try to answer the new_session before starting the record? > > On Mon, Feb 1, 2016 at 10:38 AM, Leonardo Lima Ribeiro > wrote: > Thanks Chad and Oz. > > I tried two things: > > 1) Oz suggestion: > > local new_session = freeswitch.Session("[origination_caller_id_name='7136694967 ',origination_caller_id_number='7136694967 ']sofia/gateway/4_NEXTIVA/3157244022 ", session); > new_session:setVariable(?RECORD_STEREO?,?false?) > new_session:sleep("5000") > new_session:streamFile("voicemail/vm-goodbye.wav") > new_session:sleep("10000") > new_session:hangup() > > Did not work? > > 2) Chad suggestion: > > local new_session = freeswitch.Session("[origination_caller_id_name='7136694967 ',origination_caller_id_number='7136694967 ']sofia/gateway/4_NEXTIVA/3157244022 ", session); > > uuid = new_session:get_uuid() > api = freeswitch.API() > reply = api:executeString("uuid_record " .. uuid .. " start /usr/local/freeswitch/recordings/myrecording.wav") > > new_session:sleep("5000") > new_session:streamFile("voicemail/vm-goodbye.wav") > new_session:sleep("10000") > new_session:hangup() > > Did not work too? > > Both cases I have 5 seconds of silence (first 5000 pause) and then when I play the Good Bye wav file it starts to record normally. So I have 5 seconds of silence, good bye sound and 10 seconds of recording. > > Chad, I can?t use session:recordFile because it?s a synchronous command that blocks my script.. I need to do a lot of actions like pauses, stream files etc., while the call is still going on? > > Maybe should I open a ticket for FreeSWITCH support? (I even don?t know if it exists hehe, maybe a github issue?) > > Thank you, > > >> On Jan 30, 2016, at 12:36 AM, Chad Phillips > wrote: >> >> have you tried the session method specifically for recording files? >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile >> >> or if that doesn't work, maybe this: >> >> local new_session = freeswitch.Session("someoriginatestring", session); >> uuid = new_session:get_uuid() >> api = freeswitch.API() >> reply = api:executeString("uuid_record " .. uuid .. " start " .. filepath) >> >> On Fri, Jan 29, 2016 at 1:31 PM, Oz Mortimer > wrote: >> since this is a single left call try setting record_stereo to false - https://wiki.freeswitch.org/wiki/Variable_RECORD_STEREO >> I'm probably wrong, but worth a shot! >> >> On 29 Jan 2016, at 19:53, Leonardo Ribeiro > wrote: >> >>> Hello Guys, >>> >>> Any idea? >>> I could not evolve this yet... >>> >>> Thank you, >>> >>> 2016-01-28 18:52 GMT-02:00 Leonardo Lima Ribeiro >: >>> Hello all, >>> >>> I?m trying to record an IVR using my gateway to do the outbound call in my luascript: >>> >>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 ", session); >>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav") >>> new_session:sleep("10000") >>> new_session:hangup() >>> >>> So in the above script I just call to the Bank Of America as an example and try to record the first 10 seconds of the call in the recordings path. >>> >>> The problem is that I have an empty recording file.. Why? >>> >>> The funny thing is: if I add this command after the record_session command: >>> new_session:streamFile("voicemail/vm-goodbye.wav?); >>> >>> And then this is the entire new script: >>> local new_session = freeswitch.Session("[origination_caller_id_name=?987654321',origination_caller_id_number=?987654321']sofia/gateway/MY_GATEWAY/3157244022 ", session); >>> new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav?) >>> new_session:streamFile("voicemail/vm-goodbye.wav?); >>> new_session:sleep("10000") >>> new_session:hangup() >>> >>> I can hear the Good Bye sound from my script and then hear the Bank of America IVR. >>> >>> I just don?t understand why the record works if I play a sound in our side and the record does not work if I don?t play any sound. >>> >>> Do you know what?s happening? How can I solve this? >>> >>> Thank you! >>> >>> >>> >>> >>> -- >>> Leonardo Lima Ribeiro >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/0f53b093/attachment-0001.html From chad at apartmentlines.com Mon Feb 1 19:46:01 2016 From: chad at apartmentlines.com (Chad Phillips) Date: Mon, 1 Feb 2016 09:46:01 -0700 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> Message-ID: In my profitbricks/videoconference scenario, I've been going with 10GB memory, and that's been fine. i might even be able to get away with less. you can probably find the CPU specs on profitbricks website, or certainly by contacting them, i don't know them offhand. On Mon, Feb 1, 2016 at 9:26 AM, Bote Man wrote: > It seems to me that the XML dialplan routing process would consume more > cpu than a short and simple curl request to a database server, no? > > > > Anyway, there are probably many things that can be done to streamline > performance for a particular application. I know on Windoze it was once > suggested to me a long time ago to disable the operating system updates of > file modification time stamps, for example. That can add up when a busy > system is writing to log files and creating individual CDR files to be read > and deleted by an accounting process. I don?t know how this would apply to > linux, but it demonstrates how far out of the box one can look for > performance improvements. > > > > In any case, when discussing virtualization performance it is essential to > provide specifics of the instance that runs FreeSWITCH. If someone reports > that FS ran very poorly, but does not say that it was a tiny instance that > was starved for resources then we can?t evaluate that report fairly. > > > > So the specifics in the report by Volodymyr are useful. Please keep them > coming! I am compiling these anecdotes on Confluence for others to read in > the future. > > > > Thanks. > > > > Bote > > > > > > *From:* E. Schmidbauer > *Sent:* Monday, 01 February, 2016 10:38 > *Subject:* Re: [Freeswitch-users] FreeSWITCH in virtual environments > > > > Hi Volodymyr, > > We are using xml-curl (i dont think that should have too much affect) and > hash counters only. > > No database connections, etc... > > Thanks-- 24 threads and 8gb seems like a good start! > > Emmanuel > > > > On Mon, Feb 1, 2016 at 10:34 AM, Volodymyr Fedorov > wrote: > > Hello, it depends. > > In my case 24 threads and 8gb of ram was quite enough. But I used only > xml-dialplan and hash counters. > > > > On Mon, Feb 1, 2016 at 4:16 PM, E. Schmidbauer > wrote: > > Can anyone share the VM memory/cpu specs used in these cases? > We want to run around 300 CPS on FS (running on vmware). > > very little transcoding (if any), audio only > > How much memory/cpu should be provisioned? > I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much > memory? > > Thanks, > > E > > > > On Fri, Jan 29, 2016 at 10:00 PM, servtelar wrote: > > Thanks a lot guys for sharing this info. It?s really helpful. > > > > > > On Jan 28, 2016, at 6:18 PM, Sergey Safarov wrote: > > > > We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load. > > > > Sergey > > > > On Fri, Jan 29, 2016 at 2:01 AM, servtelar wrote: > > Hi Chad > > > > How many legs you are handling with 20 cores on a conference? > > > > Regards > > > > Gustavo > > > > On Jan 28, 2016, at 7:55 PM, Chad Phillips > wrote: > > > > I've had very good luck running the newer video branch code on > ProfitBricks: https://www.profitbricks.com/ > > > > As far as I understand, the CPU cycles are guaranteed on their platform. > I've had to put as many as 20 cores on a server to handle some of our > busier video conference calls, but with that it runs quite smoothly. > > > > On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards > wrote: > > I am reviewing the Confluence Virtualization page and had some questions, > in particular about VMWare. My company distributes some of its software as > a VMWare image file and we were looking to distribute a new product using > FS in the same manner. The products operate at a customer premise, on their > VMWare infrastructure, not in a cloud environment. Since our customers > already have VMWare, switching to a different VM infrastructure is going to > hurt, so I am looking for options/alternatives. > > First, does anybody know if the virtual timing issues with VMWare have > improved since this page was last updated in 2014? Is VMWare still not good > enough? Is it possible to throw CPU & memory at this and make VMWare good > enough, or is the virtual timing just not workable? > > On the virtualization page, there was a comment from 2010 that you might > be happy with a High CPU Medium instance on AWS EC2. Certainly workload is > a factor here, but I am trying to get my head around how big a machine to > perform how small a workload. Is there a place where people talk about > their experiences? > > Are there other VM platforms that might be acceptable? > > Any help or comment is appreciated. > > Thank you, > Dan > > > _________________________________________________________________________ > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/efea40af/attachment.html From diriver63 at gmail.com Tue Feb 2 02:09:16 2016 From: diriver63 at gmail.com (Diego Rivera) Date: Mon, 1 Feb 2016 17:09:16 -0600 Subject: [Freeswitch-users] T.38 US providers Message-ID: Hello all, We use Freeswitch as our main fax server with a T38 trunk from Babytel, but now we're looking for alternatives on other reliable T38 providers in the US... mostly to see if we can get a comparable success rate at a better price. We typically fax in and out about 250k pages a week... What other good T.38 trunk providers do you guys know/recommend? Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/c5d2a16b/attachment.html From gregor at infomedia.si Tue Feb 2 01:37:23 2016 From: gregor at infomedia.si (Gregor) Date: Mon, 1 Feb 2016 22:37:23 +0000 (UTC) Subject: [Freeswitch-users] TCP registrations Message-ID: I think I am missing something. I would like to configure freeswitch that listens on TCP port for client registrations (internal profile). As I read, freeswitch should do this by default. But freeswitch responses only on UDP protocol. Is there a conf setting for specify also tcp for registrations. From sandeep.goje at gmail.com Mon Feb 1 14:14:42 2016 From: sandeep.goje at gmail.com (sandeep goje) Date: Mon, 1 Feb 2016 16:44:42 +0530 Subject: [Freeswitch-users] Sending Data in SIP BYE message Message-ID: Hi, I have a issue while setting the sip_bye_h_ headers. Here is the scenario A-->B A is a sip end point and B is a PSTN call placed through the dialplan. In this scenario, when B hangs up the call, I get proper Bye message sent to sip_bye_h headers set in the Bye message.But when A hangs up, the sip_bye_h headers are not sent in the 200 OK message. Is there a way to send the sip_bye_h headers in both BYE and 200 OK messages. I am trying to send bridge_channel or channel_name from B to A. Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160201/142dc4c6/attachment-0001.html From bote_radio at botecomm.com Tue Feb 2 03:32:59 2016 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 1 Feb 2016 19:32:59 -0500 Subject: [Freeswitch-users] TCP registrations In-Reply-To: References: Message-ID: <001b01d15d51$455900d0$d00b0270$@botecomm.com> FreeSWITCH uses UDP by default for SIP signaling. You can change this in the SIP_profile I believe. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Gregor > Sent: Monday, 01 February, 2016 17:37 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] TCP registrations > > I think I am missing something. > > I would like to configure freeswitch that listens on TCP port for client > registrations (internal profile). As I read, freeswitch should do this by > default. But freeswitch responses only on UDP protocol. Is there a conf > setting for specify also tcp for registrations. > > > __________________________________________________________ From s.safarov at gmail.com Tue Feb 2 06:07:52 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 02 Feb 2016 03:07:52 +0000 Subject: [Freeswitch-users] recording problems with mod_shout debian/master In-Reply-To: References: Message-ID: I has same case FS-8686. On Mon, Feb 1, 2016, 16:49 Luis Azedo wrote: > Hi, > > anyone having problems recording in mp3 format ? > > 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD > <96722133-5060-508 at BJC.BGI.CG.BD> start > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)* > 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match > requested rate 16000* > 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 Attaching > BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com > <995582142 at teste.sip.90e9.com>* > 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence > detection configured; assuming start of speech* > 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits > (http://lame.sf.net )* > 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass > filter disabled* > 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* > 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* > 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* > 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* > 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* > 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* > 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* > 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* > 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* > 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing > /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/ec03d8db/attachment.html From s.safarov at gmail.com Tue Feb 2 06:57:16 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 02 Feb 2016 03:57:16 +0000 Subject: [Freeswitch-users] recording problems with mod_shout debian/master In-Reply-To: References: Message-ID: Luisu you has additional info of bug. Could you reproduce this on testbox and made core dump? For me is interested core at switch_core_file.c:323 On Tue, Feb 2, 2016, 06:07 Sergey Safarov wrote: > I has same case FS-8686. > > On Mon, Feb 1, 2016, 16:49 Luis Azedo > wrote: > >> Hi, >> >> anyone having problems recording in mp3 format ? >> >> 2016-02-01 13:41:47.108078 [INFO] kazoo_node.c:625 exec: *uuid_record(96722133-5060-508 at BJC.BGI.CG.BD >> <96722133-5060-508 at BJC.BGI.CG.BD> start >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 600)* >> 2016-02-01 13:41:47.108078 [DEBUG]* switch_core_file.c:323 File >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3 sample rate 8000 doesn't match >> requested rate 16000* >> 2016-02-01 13:41:47.128429 [DEBUG] *switch_core_media_bug.c:828 >> Attaching BUG to sofia/sipinterface_1/995582142 at teste.sip.90e9.com >> <995582142 at teste.sip.90e9.com>* >> 2016-02-01 13:41:47.148020 [DEBUG] *switch_ivr_async.c:1491 No silence >> detection configured; assuming start of speech* >> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 LAME 3.99.5 64bits >> (http://lame.sf.net )* >> 2016-02-01 13:41:47.168888 [INFO] *mod_shout.c:326 polyphase lowpass >> filter disabled* >> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* >> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* >> 2016-02-01 13:41:47.168888 [ERR] *mod_shout.c:1057 MP3 encode error -1!* >> 2016-02-01 13:41:47.168888 [ERR] *switch_ivr_async.c:1160 Error writing >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* >> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* >> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* >> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* >> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* >> 2016-02-01 13:41:47.688430 [ERR] *mod_shout.c:1057 MP3 encode error -1!* >> 2016-02-01 13:41:47.688430 [ERR] *switch_ivr_async.c:1160 Error writing >> /tmp/878202b3a78863ec40f2dbc800fcde66.mp3* >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/7c3ba887/attachment.html From denis at ringme.ru Tue Feb 2 11:43:25 2016 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Tue, 2 Feb 2016 11:43:25 +0300 Subject: [Freeswitch-users] notes about amazon AWS? Message-ID: <56B06C2D.5010505@ringme.ru> Hello, who can share AMI for freeswitch and maybe some notes about setup (best practices)? But need centos 6 x86-64, it's _requirement_ some other our software. PV or HWM mode? https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 - too few info about centos. From ssinyagin at gmail.com Tue Feb 2 12:06:31 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 2 Feb 2016 10:06:31 +0100 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> Message-ID: by the way, are there any known concerns in running FreeSWITCH inside an LXC container? LXC is really convenient when network separation is required. From s.safarov at gmail.com Tue Feb 2 12:56:25 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 2 Feb 2016 12:56:25 +0300 Subject: [Freeswitch-users] notes about amazon AWS? In-Reply-To: <56B06C2D.5010505@ringme.ru> References: <56B06C2D.5010505@ringme.ru> Message-ID: PV mode not supported for new AMI. Use only HWM On Tue, Feb 2, 2016 at 11:43 AM, ????? wrote: > Hello, who can share AMI for freeswitch and maybe some notes about setup > (best practices)? But need centos 6 x86-64, it's _requirement_ some > other our software. > > PV or HWM mode? > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 - too > few info about centos. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/a40c3116/attachment-0001.html From stephen.thwaites at callstera.com Tue Feb 2 13:05:14 2016 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Tue, 2 Feb 2016 11:05:14 +0100 Subject: [Freeswitch-users] BLF Subscriptions sometimes don't send an initial Notify Message-ID: Hello, I have setup presence and in most cases it is working as expected. i.e. Subscription is sent to FS, FS returns Accepted then immediately FS sends the notify to the phone, thereafter all Notifies for ringing, pickup and hangup. Great. However in some cases a subscribe to an extension does subscribe, FS sends the accepted response but a Notify is not sent out at that point. However if I call the extension the Notify works perfectly. Any ideas of what could cause the initial Notify not to be sent after the Acceptance 202? Any help would be appreciated. Regards, Steve. Some info below: FS is configured as Multi-Tennant ** Multi-Tennant SIP Trace for Subscription FS Receives this from the phone: SUBSCRIBE sip:203 at xxx.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport From: ;tag=946578510 To: ;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.5.29 Expires: 480 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 FS Sends this back to the phone: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.24:25060 ;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x From: ;tag=946578510 To: ;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: Expires: 480 User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=480 Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/106d202d/attachment.html From lists at telefaks.de Tue Feb 2 14:40:37 2016 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 02 Feb 2016 12:40:37 +0100 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> Message-ID: <56B095B5.8090804@telefaks.de> Hello Stanislav, we have just setup Freeswitch on Debian 8 inside LXC. We are examinating to switch from OpenVZ zu LXC. Freeswitch works so far productive, but with almost no load, so I have no performance figures yet. We will do some load and stress tests during this month to compare against OpenVZ. We expect advantages compared to OpenVZ with a newer kernel and in networking, let's see then where the drawbacks are. Best regards Peter On 02/02/16 10:06, Stanislav Sinyagin wrote: > by the way, are there any known concerns in running FreeSWITCH inside > an LXC container? > > LXC is really convenient when network separation is required. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From deforceczt at gmail.com Tue Feb 2 15:08:49 2016 From: deforceczt at gmail.com (Vladislav Ivanov) Date: Tue, 2 Feb 2016 14:08:49 +0200 Subject: [Freeswitch-users] Does freeswitch forks his processes? Message-ID: Hey guys, I have a question about freeswitch process/threading usage. So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance. http://i.imgur.com/bdbYOwp.png But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run: http://i.imgur.com/VNpl55z.jpg I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration. Be it physical or virtual environment I cant pass the 50 cps mark. I have strange issue with CPU usage on same CPS: http://i.imgur.com/8BdQWVL.png http://i.imgur.com/mWRnoGr.png I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere. I have tested configuration on: Debian 8 2 core/8 gb ram 4 core/8 gb ram (graphs are from here) 8 core/32 gb ram and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy. Test is: sipp -> freeswitch -> sipp Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile... Any advice? Thank you all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/bcdee733/attachment.html From gregor at infomedia.si Tue Feb 2 11:50:14 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 2 Feb 2016 09:50:14 +0100 Subject: [Freeswitch-users] TCP registrations In-Reply-To: <001b01d15d51$455900d0$d00b0270$@botecomm.com> References: <001b01d15d51$455900d0$d00b0270$@botecomm.com> Message-ID: Yes, I also think so, but cannot find explicitly documented. So please, if anyone know exactly which command is, please help. 2016-02-02 1:32 GMT+01:00 Bote Man : > FreeSWITCH uses UDP by default for SIP signaling. You can change this in > the > SIP_profile I believe. > > > --- > Bote > > FreeSWITCH Docs Janitor > http://freeswitch.org/confluence > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > > users-bounces at lists.freeswitch.org] On Behalf Of Gregor > > Sent: Monday, 01 February, 2016 17:37 > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] TCP registrations > > > > I think I am missing something. > > > > I would like to configure freeswitch that listens on TCP port for client > > registrations (internal profile). As I read, freeswitch should do this by > > default. But freeswitch responses only on UDP protocol. Is there a conf > > setting for specify also tcp for registrations. > > > > > > __________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/a005103e/attachment.html From roman at dissauer.net Tue Feb 2 14:58:22 2016 From: roman at dissauer.net (Roman Dissauer) Date: Tue, 2 Feb 2016 12:58:22 +0100 Subject: [Freeswitch-users] BLF for Gateway Message-ID: Hi All, is there a way to get gateway usage in freeswitch on my phones blf? I have multiple gateways registered and want to see which one is taken for a particular outbound call. Best Regards, Roman From bote_radio at botecomm.com Tue Feb 2 15:49:17 2016 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 2 Feb 2016 07:49:17 -0500 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: <56B095B5.8090804@telefaks.de> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> <56B095B5.8090804@telefaks.de> Message-ID: <005601d15db8$219762b0$64c62810$@botecomm.com> Peter, I / we would be grateful if you would kindly update us with your experiences on this page https://freeswitch.org/confluence/display/FREESWITCH/Virtualization+Experien ces as well as the mailing list. Because each installation is different it is helpful to compile configurations that work and don't work where others can benefit from these experiences. Vielen Dank! --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence > -----Original Message----- > From: Peter Steinbach > Sent: Tuesday, 02 February, 2016 06:41 > Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments > > Hello Stanislav, > > we have just setup Freeswitch on Debian 8 inside LXC. We are examinating > to switch from OpenVZ zu LXC. > > Freeswitch works so far productive, but with almost no load, so I have > no performance figures yet. We will do some load and stress tests during > this month to compare against OpenVZ. > > We expect advantages compared to OpenVZ with a newer kernel and in > networking, let's see then where the drawbacks are. > > Best regards > Peter > > On 02/02/16 10:06, Stanislav Sinyagin wrote: > > by the way, are there any known concerns in running FreeSWITCH inside > > an LXC container? > > > > LXC is really convenient when network separation is required. > > > > > __________________________________________________________ > _______________ From lists at plustel.dk Tue Feb 2 15:21:43 2016 From: lists at plustel.dk (Tom Braarup Cuykens) Date: Tue, 2 Feb 2016 13:21:43 +0100 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: <56B095B5.8090804@telefaks.de> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> <007701d15d0d$4dfba330$e9f2e990$@botecomm.com> <56B095B5.8090804@telefaks.de> Message-ID: <56B09F57.2010801@plustel.dk> Hello Peter, This is very interesting. Would love to have feedback and recipe to see how it worked out. Kind Regards, Tom Braarup Cuykens On 02/02/2016 12:40 PM, Peter Steinbach wrote: > Hello Stanislav, > > we have just setup Freeswitch on Debian 8 inside LXC. We are examinating > to switch from OpenVZ zu LXC. > > Freeswitch works so far productive, but with almost no load, so I have > no performance figures yet. We will do some load and stress tests during > this month to compare against OpenVZ. > > We expect advantages compared to OpenVZ with a newer kernel and in > networking, let's see then where the drawbacks are. > > Best regards > Peter > > On 02/02/16 10:06, Stanislav Sinyagin wrote: >> by the way, are there any known concerns in running FreeSWITCH inside >> an LXC container? >> >> LXC is really convenient when network separation is required. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From royj at yandex.ru Tue Feb 2 15:54:43 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Tue, 02 Feb 2016 15:54:43 +0300 Subject: [Freeswitch-users] async, sendmsg, execute-app-name Message-ID: <4661454417683@web12o.yandex.ru> Hi, all I am working on nodejs outbound socket application and faced that if send to FreeSWITCH message like: sendmsg execute-app-name: playback execute-app-arg: silence_stream://2000 call-command: execute when there is already no corresponding channel (a caller hung up by itself and from FreesSWITCH received CHANNEL_HANGUP_COMPLETE, text/disconnect-notice;linger) FreesSWITCH answers: Content-Type: command/reply Reply-Text: +OK Logic of library and application such, that we know that an application is finished, when received CHANNEL_EXECUTE_COMPLETE. There are cases when a caller hung up and exactly in this moment application not yet knows about it and sends message 'call-command: execute' and then there is javascript Promise in pending state, seems like forever. Is there any chance to understand in the answer that the channel does not exist? From bote_radio at botecomm.com Tue Feb 2 15:58:46 2016 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 2 Feb 2016 07:58:46 -0500 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: Message-ID: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime threads. If not started as root, this would explain your CPS limitations. There are also limits that can be set in the config files. After it starts it drops privileges to those specified on the command line with ?u and ?g switches. FreeSWITCH uses multi-threading. I do not know about htop, but maybe it is showing the multiple threads? top ?H shows each thread. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: Vladislav Ivanov Sent: Tuesday, 02 February, 2016 07:09 Subject: [Freeswitch-users] Does freeswitch forks his processes? Hey guys, I have a question about freeswitch process/threading usage. So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance. http://i.imgur.com/bdbYOwp.png But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run: http://i.imgur.com/VNpl55z.jpg I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration. Be it physical or virtual environment I cant pass the 50 cps mark. I have strange issue with CPU usage on same CPS: http://i.imgur.com/8BdQWVL.png http://i.imgur.com/mWRnoGr.png I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere. I have tested configuration on: Debian 8 2 core/8 gb ram 4 core/8 gb ram (graphs are from here) 8 core/32 gb ram and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy. Test is: sipp -> freeswitch -> sipp Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile... Any advice? Thank you all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/9e9abcff/attachment.html From giacomo.vacca at gmail.com Tue Feb 2 15:59:19 2016 From: giacomo.vacca at gmail.com (Giacomo Vacca) Date: Tue, 2 Feb 2016 13:59:19 +0100 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: Message-ID: FreeSWITCH is a single process, multi-threading application. htop is showing you threads activity. On 2 February 2016 at 13:08, Vladislav Ivanov wrote: > Hey guys, > > I have a question about freeswitch process/threading usage. > So far that I haven't noticed freeswitch to fork himself, I have only 1 > freeswitch instance. > http://i.imgur.com/bdbYOwp.png > > But then I found screenshot of htop with freeswitch and noticed that there > is multiple freeswitch processes being run: > http://i.imgur.com/VNpl55z.jpg > > I'm having issues with "loading" the freeswitch after 50 cps in any > cpu/ram configuration. > Be it physical or virtual environment I cant pass the 50 cps mark. > I have strange issue with CPU usage on same CPS: > > http://i.imgur.com/8BdQWVL.png > http://i.imgur.com/mWRnoGr.png > > I timeload test freeswitch with 50cps for 5+ hours, and seems like there > is some kind of leak somewhere. > I have tested configuration on: > Debian 8 > 2 core/8 gb ram > 4 core/8 gb ram (graphs are from here) > 8 core/32 gb ram > > and in all the tests I were not able to send more than 50 cps without CPU > dropping to 0 with all system starting to respond really laggy. > > Test is: > sipp -> freeswitch -> sipp > > Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and > 1 profile... > Any advice? > > Thank you all > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/5f96b594/attachment.html From s.safarov at gmail.com Tue Feb 2 16:01:23 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 2 Feb 2016 16:01:23 +0300 Subject: [Freeswitch-users] TCP registrations In-Reply-To: References: <001b01d15d51$455900d0$d00b0270$@botecomm.com> Message-ID: FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml also TLS transport is will be enabled. To check what is type of socket is open on server please use netstat -an --inet | grep -P "5060|5061|5080" Example [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080" tcp 0 0 217.12.247.214:5060 0.0.0.0:* LISTEN tcp 0 0 10.21.7.30:5060 0.0.0.0:* LISTEN tcp 0 0 217.12.247.214:5061 0.0.0.0:* LISTEN tcp 0 0 217.12.247.214:5080 0.0.0.0:* LISTEN udp 0 0 217.12.247.214:5060 0.0.0.0:* udp 0 0 10.21.7.30:5060 0.0.0.0:* udp 0 0 217.12.247.214:5080 0.0.0.0:* On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger wrote: > Yes, I also think so, but cannot find explicitly documented. So please, if > anyone know exactly which command is, please help. > > 2016-02-02 1:32 GMT+01:00 Bote Man : > >> FreeSWITCH uses UDP by default for SIP signaling. You can change this in >> the >> SIP_profile I believe. >> >> >> --- >> Bote >> >> FreeSWITCH Docs Janitor >> http://freeswitch.org/confluence >> >> >> >> >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- >> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor >> > Sent: Monday, 01 February, 2016 17:37 >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: [Freeswitch-users] TCP registrations >> > >> > I think I am missing something. >> > >> > I would like to configure freeswitch that listens on TCP port for client >> > registrations (internal profile). As I read, freeswitch should do this >> by >> > default. But freeswitch responses only on UDP protocol. Is there a conf >> > setting for specify also tcp for registrations. >> > >> > >> > __________________________________________________________ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 > ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia > ? www.infomedia.si > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/9c20d415/attachment-0001.html From asilva at wirelessmundi.com Tue Feb 2 16:19:14 2016 From: asilva at wirelessmundi.com (Antonio Silva) Date: Tue, 2 Feb 2016 14:19:14 +0100 Subject: [Freeswitch-users] TCP registrations In-Reply-To: References: <001b01d15d51$455900d0$d00b0270$@botecomm.com> Message-ID: <56B0ACD2.9070800@wirelessmundi.com> The parameter is "bind-params" https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files by default fs should bind to tcp and udp but if you want only tcp just set for the profile: On 02/02/2016 02:01 PM, Sergey Safarov wrote: > FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml > also TLS transport is will be enabled. > To check what is type of socket is open on server please use > netstat -an --inet | grep -P "5060|5061|5080" > > Example > [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080" > tcp 0 0 217.12.247.214:5060 > 0.0.0.0:* LISTEN > tcp 0 0 10.21.7.30:5060 0.0.0.0:* > LISTEN > tcp 0 0 217.12.247.214:5061 > 0.0.0.0:* LISTEN > tcp 0 0 217.12.247.214:5080 > 0.0.0.0:* LISTEN > udp 0 0 217.12.247.214:5060 > 0.0.0.0:* > udp 0 0 10.21.7.30:5060 0.0.0.0:* > udp 0 0 217.12.247.214:5080 > 0.0.0.0:* > > On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger > wrote: > > Yes, I also think so, but cannot find explicitly documented. So > please, if anyone know exactly which command is, please help. > > 2016-02-02 1:32 GMT+01:00 Bote Man >: > > FreeSWITCH uses UDP by default for SIP signaling. You can > change this in the > SIP_profile I believe. > > > --- > Bote > > FreeSWITCH Docs Janitor > http://freeswitch.org/confluence > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch- > > users-bounces at lists.freeswitch.org > ] On Behalf Of Gregor > > Sent: Monday, 01 February, 2016 17:37 > > To: freeswitch-users at lists.freeswitch.org > > > Subject: [Freeswitch-users] TCP registrations > > > > I think I am missing something. > > > > I would like to configure freeswitch that listens on TCP > port for client > > registrations (internal profile). As I read, freeswitch > should do this by > > default. But freeswitch responses only on UDP protocol. Is > there a conf > > setting for specify also tcp for registrations. > > > > > > __________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Gregor Nanger > *CTO* > t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 > ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia > ? www.infomedia.si > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos, Ant?nio silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/6b6db78c/attachment.html From gb at cm.nl Tue Feb 2 16:37:28 2016 From: gb at cm.nl (Grant Bagdasarian) Date: Tue, 2 Feb 2016 13:37:28 +0000 Subject: [Freeswitch-users] FreeSWITCH in virtual environments In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C9979B74@PHXEX2.vertical.com> <88218521-FB01-4698-9DBF-625D99E8BC79@gmail.com> <58E2C5C4-87FF-4E01-8048-6C39B9362B02@gmail.com> Message-ID: <1171818c8a824ad2ad7f37a0688f39a2@CM-EX-V01.cm.local> We never tested with 300 CPS, but the host itself has 8 GB of memory. Our VM?s usually have 2-4 CPU?s and 1-2 GB of memory, but memory is hardly used. It?s CPU you need. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of E. Schmidbauer Sent: Monday, February 1, 2016 4:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH in virtual environments Can anyone share the VM memory/cpu specs used in these cases? We want to run around 300 CPS on FS (running on vmware). very little transcoding (if any), audio only How much memory/cpu should be provisioned? I see Grant mentioned "single VM host (6 Cores, 12 Threads)" but how much memory? Thanks, E On Fri, Jan 29, 2016 at 10:00 PM, servtelar > wrote: Thanks a lot guys for sharing this info. It?s really helpful. On Jan 28, 2016, at 6:18 PM, Sergey Safarov > wrote: We have to core ESXi vm with 140 session (70 calls) with have 70 CPU load. Sergey On Fri, Jan 29, 2016 at 2:01 AM, servtelar > wrote: Hi Chad How many legs you are handling with 20 cores on a conference? Regards Gustavo On Jan 28, 2016, at 7:55 PM, Chad Phillips > wrote: I've had very good luck running the newer video branch code on ProfitBricks: https://www.profitbricks.com/ As far as I understand, the CPU cycles are guaranteed on their platform. I've had to put as many as 20 cores on a server to handle some of our busier video conference calls, but with that it runs quite smoothly. On Thu, Jan 28, 2016 at 2:15 PM, Dan Edwards > wrote: I am reviewing the Confluence Virtualization page and had some questions, in particular about VMWare. My company distributes some of its software as a VMWare image file and we were looking to distribute a new product using FS in the same manner. The products operate at a customer premise, on their VMWare infrastructure, not in a cloud environment. Since our customers already have VMWare, switching to a different VM infrastructure is going to hurt, so I am looking for options/alternatives. First, does anybody know if the virtual timing issues with VMWare have improved since this page was last updated in 2014? Is VMWare still not good enough? Is it possible to throw CPU & memory at this and make VMWare good enough, or is the virtual timing just not workable? On the virtualization page, there was a comment from 2010 that you might be happy with a High CPU Medium instance on AWS EC2. Certainly workload is a factor here, but I am trying to get my head around how big a machine to perform how small a workload. Is there a place where people talk about their experiences? Are there other VM platforms that might be acceptable? Any help or comment is appreciated. Thank you, Dan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/2d3ba0fe/attachment-0001.html From krice at freeswitch.org Tue Feb 2 17:53:10 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 2 Feb 2016 08:53:10 -0600 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: Message-ID: <555b01d15dc9$70137cb0$503a7610$@freeswitch.org> FreeSWITCH uses threading? in modern linux kernels threading and forking are very similar? if you look in top you?ll only see 1 FS process that?s because top by default rolls up all the threads? htop on the other hand by default shows you the individual threads. In FreeSWITCH there are several threads running on just a base idle FreeSWITCH process. Each addition call leg is atleast 1 more thread. Depending on which applications are actually active on a call, there could be more then 1 thread per call leg. On the load testing and load handling capabilities this is something the FS team typically does not support on the open source side. I would recommend contacting consulting at freeswitch.org to get the pros involved. There are a lot of factors to take into consideration and just 1 param can make orders of magnitude difference from test to test K From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladislav Ivanov Sent: Tuesday, February 2, 2016 6:09 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Does freeswitch forks his processes? Hey guys, I have a question about freeswitch process/threading usage. So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance. http://i.imgur.com/bdbYOwp.png But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run: http://i.imgur.com/VNpl55z.jpg I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration. Be it physical or virtual environment I cant pass the 50 cps mark. I have strange issue with CPU usage on same CPS: http://i.imgur.com/8BdQWVL.png http://i.imgur.com/mWRnoGr.png I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere. I have tested configuration on: Debian 8 2 core/8 gb ram 4 core/8 gb ram (graphs are from here) 8 core/32 gb ram and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy. Test is: sipp -> freeswitch -> sipp Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile... Any advice? Thank you all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/bff9bc33/attachment.html From lists at telefaks.de Tue Feb 2 18:10:54 2016 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 02 Feb 2016 16:10:54 +0100 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> Message-ID: <56B0C6FE.9070507@telefaks.de> I've just stumpled over this: >Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime threads. If not started as root, this would explain your CPS limitations. We like to run Freeswitch as a non privileged user, due to security concerns. So there are drawbacks here compared to running FS as root? Can we somehow quantify the differences? Best regards Peter On 02/02/16 13:58, Bote Man wrote: > > Is FreeSWITCH starting with root permissions? It needs this in order > to use the FIFO scheduler and access realtime threads. If not started > as root, this would explain your CPS limitations. There are also > limits that can be set in the config files. > > > > After it starts it drops privileges to those specified on the command > line with --u and --g switches. > > > > FreeSWITCH uses multi-threading. I do not know about htop, but maybe > it is showing the multiple threads? > > > > top --H shows each thread. > > > > --- > > Bote > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > > > *From:*Vladislav Ivanov > *Sent:* Tuesday, 02 February, 2016 07:09 > *Subject:* [Freeswitch-users] Does freeswitch forks his processes? > > > > Hey guys, > > I have a question about freeswitch process/threading usage. > So far that I haven't noticed freeswitch to fork himself, I have only > 1 freeswitch instance. > http://i.imgur.com/bdbYOwp.png > > But then I found screenshot of htop with freeswitch and noticed that > there is multiple freeswitch processes being run: > http://i.imgur.com/VNpl55z.jpg > > I'm having issues with "loading" the freeswitch after 50 cps in any > cpu/ram configuration. > Be it physical or virtual environment I cant pass the 50 cps mark. > I have strange issue with CPU usage on same CPS: > > http://i.imgur.com/8BdQWVL.png > http://i.imgur.com/mWRnoGr.png > > I timeload test freeswitch with 50cps for 5+ hours, and seems like > there is some kind of leak somewhere. > I have tested configuration on: > Debian 8 > 2 core/8 gb ram > 4 core/8 gb ram (graphs are from here) > 8 core/32 gb ram > > and in all the tests I were not able to send more than 50 cps without > CPU dropping to 0 with all system starting to respond really laggy. > > Test is: > sipp -> freeswitch -> sipp > > Just 1 dialpeer with bridge action. No gateways. Just simple dialplan > and 1 profile... > Any advice? > > Thank you all > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/08c37267/attachment.html From nneul at mst.edu Tue Feb 2 18:22:31 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 2 Feb 2016 09:22:31 -0600 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: <56B0C6FE.9070507@telefaks.de> References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> Message-ID: <56B0C9B7.8000106@mst.edu> "Run as" != "Start as" If you insist on not starting FS as root to let it change user, like most other daemons/services, you'll have to jump through a bunch of extra steps using file system capabilities to give it the ability to set scheduler parameters/etc that are restricted to root normally. -- Nathan On 02/02/2016 09:10 AM, Peter Steinbach wrote: > I've just stumpled over this: > >Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime > threads. If not started as root, this would explain your CPS limitations. > > We like to run Freeswitch as a non privileged user, due to security concerns. So there are drawbacks here compared to > running FS as root? Can we somehow quantify the differences? > > Best regards > Peter > > > On 02/02/16 13:58, Bote Man wrote: >> >> Is FreeSWITCH starting with root permissions? It needs this in order to use the FIFO scheduler and access realtime >> threads. If not started as root, this would explain your CPS limitations. There are also limits that can be set in the >> config files. >> >> After it starts it drops privileges to those specified on the command line with ?u and ?g switches. >> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe it is showing the multiple threads? >> >> top ?H shows each thread. >> >> --- >> >> Bote >> >> FreeSWITCH Docs Janitor >> >> http://freeswitch.org/confluence >> >> *From:*Vladislav Ivanov >> *Sent:* Tuesday, 02 February, 2016 07:09 >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >> >> Hey guys, >> >> I have a question about freeswitch process/threading usage. >> So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance. >> http://i.imgur.com/bdbYOwp.png >> >> But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run: >> http://i.imgur.com/VNpl55z.jpg >> >> I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration. >> Be it physical or virtual environment I cant pass the 50 cps mark. >> I have strange issue with CPU usage on same CPS: >> >> http://i.imgur.com/8BdQWVL.png >> http://i.imgur.com/mWRnoGr.png >> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere. >> I have tested configuration on: >> Debian 8 >> 2 core/8 gb ram >> 4 core/8 gb ram (graphs are from here) >> 8 core/32 gb ram >> >> and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to >> respond really laggy. >> >> Test is: >> sipp -> freeswitch -> sipp >> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile... >> Any advice? >> >> Thank you all >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet:www.telefaks.de > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From brian at freeswitch.org Tue Feb 2 20:07:59 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Feb 2016 11:07:59 -0600 Subject: [Freeswitch-users] TCP registrations In-Reply-To: <56B0ACD2.9070800@wirelessmundi.com> References: <001b01d15d51$455900d0$d00b0270$@botecomm.com> <56B0ACD2.9070800@wirelessmundi.com> Message-ID: I'm going to guess your device probably fails to send the transport=tcp on the contact there for it probably registers over TCP but we contact it back over UDP? Can you confirm? On Tue, Feb 2, 2016 at 7:19 AM, Antonio Silva wrote: > The parameter is "bind-params" > > > https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files > > by default fs should bind to tcp and udp but if you want only tcp just set > for the profile: > > > > > > > On 02/02/2016 02:01 PM, Sergey Safarov wrote: > > FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml > also TLS transport is will be enabled. > To check what is type of socket is open on server please use > netstat -an --inet | grep -P "5060|5061|5080" > > Example > [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080" > tcp 0 0 217.12.247.214:5060 0.0.0.0:* > LISTEN > tcp 0 0 10.21.7.30:5060 0.0.0.0:* > LISTEN > tcp 0 0 217.12.247.214:5061 0.0.0.0:* > LISTEN > tcp 0 0 217.12.247.214:5080 0.0.0.0:* > LISTEN > udp 0 0 217.12.247.214:5060 0.0.0.0:* > > udp 0 0 10.21.7.30:5060 0.0.0.0:* > > udp 0 0 217.12.247.214:5080 0.0.0.0:* > > On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger > wrote: > >> Yes, I also think so, but cannot find explicitly documented. So please, >> if anyone know exactly which command is, please help. >> >> 2016-02-02 1:32 GMT+01:00 Bote Man < >> bote_radio at botecomm.com>: >> >>> FreeSWITCH uses UDP by default for SIP signaling. You can change this in >>> the >>> SIP_profile I believe. >>> >>> >>> --- >>> Bote >>> >>> FreeSWITCH Docs Janitor >>> http://freeswitch.org/confluence >>> >>> >>> >>> >>> > -----Original Message----- >>> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch- >>> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor >>> > Sent: Monday, 01 February, 2016 17:37 >>> > To: freeswitch-users at lists.freeswitch.org >>> > Subject: [Freeswitch-users] TCP registrations >>> > >>> > I think I am missing something. >>> > >>> > I would like to configure freeswitch that listens on TCP port for >>> client >>> > registrations (internal profile). As I read, freeswitch should do this >>> by >>> > default. But freeswitch responses only on UDP protocol. Is there a conf >>> > setting for specify also tcp for registrations. >>> > >>> > >>> > __________________________________________________________ >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Gregor Nanger >> >> *CTO* >> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 >> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia >> ? www.infomedia.si >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.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-usershttp://www.freeswitch.org > > > -- > > Saludos / Regards / Cumprimentos, > Ant?nio silva > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/b35f5892/attachment.html From therebel22 at gmail.com Tue Feb 2 20:20:04 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 18:20:04 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: Hello, sorry for asking again : I hav an asterisk that register on Freeswitch (as a user). When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension. Here my bridge tests => s extension in asterisk instead of extension => Not authenticated in asterisk (because no IP authentication in asterisk) Have you an idea how to send real extension instead of s extension ? Thanks 2016-01-03 10:45 GMT+01:00 Marc S : > Hello, > > i'm discovering FS. I hav read a lot about users and gateways. > > I would like to FS act as registrar for authenticated SIP trunking. > > - Customers IPBX would register with login/password to Freeswitch. > - Incoming call would be routed to these SIP trunks in dialplan XML. > > directory/users does not seem to be the solution because in dialplan, > destination DID can't be defined, only user id : > > > > gateway seems to be designed for SIP trunking to remote SIP gateway, not > for FS to act as registrar. > > Is it possible to FS to act as authenticated SIP trunking registrar ? > > Thanks a lot, > Marc > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/7db8c993/attachment-0001.html From s.safarov at gmail.com Tue Feb 2 21:16:27 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 2 Feb 2016 21:16:27 +0300 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: What is way you planing to use for link DID with user? Sergey On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: > Hello, > sorry for asking again : > > I hav an asterisk that register on Freeswitch (as a user). > > When a call is incoming to FS, FS send it to asterisk : In asterisk, it is > the s extension. > > Here my bridge tests > > > > => s extension in asterisk instead of extension > > > > => Not authenticated in asterisk (because no IP authentication in asterisk) > > Have you an idea how to send real extension instead of s extension ? > Thanks > > > > > > > > 2016-01-03 10:45 GMT+01:00 Marc S : > >> Hello, >> >> i'm discovering FS. I hav read a lot about users and gateways. >> >> I would like to FS act as registrar for authenticated SIP trunking. >> >> - Customers IPBX would register with login/password to Freeswitch. >> - Incoming call would be routed to these SIP trunks in dialplan XML. >> >> directory/users does not seem to be the solution because in dialplan, >> destination DID can't be defined, only user id : >> >> >> >> gateway seems to be designed for SIP trunking to remote SIP gateway, not >> for FS to act as registrar. >> >> Is it possible to FS to act as authenticated SIP trunking registrar ? >> >> Thanks a lot, >> Marc >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/f3ed474e/attachment.html From therebel22 at gmail.com Tue Feb 2 22:03:34 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 20:03:34 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: I want to use mod xml curl to generate dynamic dialplan xml like this : Thanks 2016-02-02 19:16 GMT+01:00 Sergey Safarov : > What is way you planing to use for link DID with user? > > Sergey > > On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: > >> Hello, >> sorry for asking again : >> >> I hav an asterisk that register on Freeswitch (as a user). >> >> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >> is the s extension. >> >> Here my bridge tests >> >> >> >> => s extension in asterisk instead of extension >> >> >> >> => Not authenticated in asterisk (because no IP authentication in >> asterisk) >> >> Have you an idea how to send real extension instead of s extension ? >> Thanks >> >> >> >> >> >> >> >> 2016-01-03 10:45 GMT+01:00 Marc S : >> >>> Hello, >>> >>> i'm discovering FS. I hav read a lot about users and gateways. >>> >>> I would like to FS act as registrar for authenticated SIP trunking. >>> >>> - Customers IPBX would register with login/password to Freeswitch. >>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>> >>> directory/users does not seem to be the solution because in dialplan, >>> destination DID can't be defined, only user id : >>> >>> >>> >>> gateway seems to be designed for SIP trunking to remote SIP gateway, not >>> for FS to act as registrar. >>> >>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>> >>> Thanks a lot, >>> Marc >>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/963efcc1/attachment.html From omortimer at gmail.com Tue Feb 2 22:23:16 2016 From: omortimer at gmail.com (Oz Mortimer) Date: Tue, 2 Feb 2016 19:23:16 +0000 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". Google is your friend.. > On 2 Feb 2016, at 19:03, Marc S wrote: > > I want to use mod xml curl to generate dynamic dialplan xml like this : > > > > > > > > Thanks > > > > 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >> What is way you planing to use for link DID with user? >> >> Sergey >> >>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>> Hello, >>> sorry for asking again : >>> >>> I hav an asterisk that register on Freeswitch (as a user). >>> >>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension. >>> >>> Here my bridge tests >>> >>> >>> >>> => s extension in asterisk instead of extension >>> >>> >>> >>> => Not authenticated in asterisk (because no IP authentication in asterisk) >>> >>> Have you an idea how to send real extension instead of s extension ? >>> Thanks >>> >>> >>> >>> >>> >>> >>> >>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>> Hello, >>>> >>>> i'm discovering FS. I hav read a lot about users and gateways. >>>> >>>> I would like to FS act as registrar for authenticated SIP trunking. >>>> >>>> - Customers IPBX would register with login/password to Freeswitch. >>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>> >>>> directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id : >>>> >>>> >>>> >>>> gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar. >>>> >>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>> >>>> Thanks a lot, >>>> Marc >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/226ff029/attachment-0001.html From therebel22 at gmail.com Tue Feb 2 22:28:11 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 20:28:11 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: It seems that "freeswitch gateway username" return all results about setup username and password on FS to register against external SIP gateway, am i wrong ? 2016-02-02 20:23 GMT+01:00 Oz Mortimer : > Setup username / password authentication on asterisk and set the > corresponding user & pass in your freeswitch gateway. > I'm sure a google for "asterisk username authentication" and "freeswitch > gateway username" will give you plenty of examples - you will want > freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". > Google is your friend.. > > > On 2 Feb 2016, at 19:03, Marc S wrote: > > I want to use mod xml curl to generate dynamic dialplan xml like this : > > > > > > > > Thanks > > > > 2016-02-02 19:16 GMT+01:00 Sergey Safarov : > >> What is way you planing to use for link DID with user? >> >> Sergey >> >> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >> >>> Hello, >>> sorry for asking again : >>> >>> I hav an asterisk that register on Freeswitch (as a user). >>> >>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>> is the s extension. >>> >>> Here my bridge tests >>> >>> >>> >>> => s extension in asterisk instead of extension >>> >>> >>> >>> => Not authenticated in asterisk (because no IP authentication in >>> asterisk) >>> >>> Have you an idea how to send real extension instead of s extension ? >>> Thanks >>> >>> >>> >>> >>> >>> >>> >>> 2016-01-03 10:45 GMT+01:00 Marc S : >>> >>>> Hello, >>>> >>>> i'm discovering FS. I hav read a lot about users and gateways. >>>> >>>> I would like to FS act as registrar for authenticated SIP trunking. >>>> >>>> - Customers IPBX would register with login/password to Freeswitch. >>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>> >>>> directory/users does not seem to be the solution because in dialplan, >>>> destination DID can't be defined, only user id : >>>> >>>> >>>> >>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>> not for FS to act as registrar. >>>> >>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>> >>>> Thanks a lot, >>>> Marc >>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/c711c5cb/attachment.html From blessendor at gmail.com Tue Feb 2 22:57:12 2016 From: blessendor at gmail.com (Alexandr Usov) Date: Tue, 2 Feb 2016 21:57:12 +0200 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: You must have in Asterisk configs something as in my examples. ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. With static IP of both FS and Asterisk you not need to do register from asterisk. register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch ;; type=friend means that this peer can be used for outbound and inbound calls, so we don't need to create two peer settings block ( [freeswitch-in] and [freeswitch-out] ) [freeswitch] ;; you maybe want to use here your public DID number, as well as in username/fromuser settings type=friend username=asterisk2fs fromuser=asterisk2fs fromdomain=ip_of_freeswitch host=ip_of_freeswitch context=from-freeswitch secret=strongpassword insecure=port,invite qualify=yes port=5060 Your context [from-freeswitch] must have an extension, named as 'freeswitch' (or your DID number) for incoming calls operations. 2016-02-02 21:28 GMT+02:00 Marc S : > It seems that "freeswitch gateway username" return all results about setup > username and password on FS to register against external SIP gateway, am i > wrong ? > > 2016-02-02 20:23 GMT+01:00 Oz Mortimer : > >> Setup username / password authentication on asterisk and set the >> corresponding user & pass in your freeswitch gateway. >> I'm sure a google for "asterisk username authentication" and "freeswitch >> gateway username" will give you plenty of examples - you will want >> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >> Google is your friend.. >> >> >> On 2 Feb 2016, at 19:03, Marc S wrote: >> >> I want to use mod xml curl to generate dynamic dialplan xml like this : >> >> >> >> >> >> >> >> Thanks >> >> >> >> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >> >>> What is way you planing to use for link DID with user? >>> >>> Sergey >>> >>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>> >>>> Hello, >>>> sorry for asking again : >>>> >>>> I hav an asterisk that register on Freeswitch (as a user). >>>> >>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>>> is the s extension. >>>> >>>> Here my bridge tests >>>> >>>> >>>> >>>> => s extension in asterisk instead of extension >>>> >>>> >>>> >>>> => Not authenticated in asterisk (because no IP authentication in >>>> asterisk) >>>> >>>> Have you an idea how to send real extension instead of s extension ? >>>> Thanks >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>> >>>>> Hello, >>>>> >>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>> >>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>> >>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>> >>>>> directory/users does not seem to be the solution because in dialplan, >>>>> destination DID can't be defined, only user id : >>>>> >>>>> >>>>> >>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>> not for FS to act as registrar. >>>>> >>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>> >>>>> Thanks a lot, >>>>> Marc >>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/b2d6ba95/attachment-0001.html From therebel22 at gmail.com Tue Feb 2 23:06:01 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 21:06:01 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: Thanks, Asterisk is already registred to FS. But when incoming call to FS is bridged to registered Asterisk : FS send SIP Message to asterisk : INVITE s@ instead of : INVITE 12345678@ I would like to get 12345678 in asterisk.. 2016-02-02 20:57 GMT+01:00 Alexandr Usov : > You must have in Asterisk configs something as in my examples. > > ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. > With static IP of both FS and Asterisk you not need to do register from > asterisk. > register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch > > > ;; type=friend means that this peer can be used for outbound and inbound > calls, so we don't need to create two peer settings block ( [freeswitch-in] > and [freeswitch-out] ) > > [freeswitch] ;; you maybe want to use here your public DID number, as well > as in username/fromuser settings > type=friend > username=asterisk2fs > fromuser=asterisk2fs > fromdomain=ip_of_freeswitch > host=ip_of_freeswitch > context=from-freeswitch > secret=strongpassword > insecure=port,invite > qualify=yes > port=5060 > > > Your context [from-freeswitch] must have an extension, named as > 'freeswitch' (or your DID number) for incoming calls operations. > > > > > 2016-02-02 21:28 GMT+02:00 Marc S : > >> It seems that "freeswitch gateway username" return all results about >> setup username and password on FS to register against external SIP gateway, >> am i wrong ? >> >> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >> >>> Setup username / password authentication on asterisk and set the >>> corresponding user & pass in your freeswitch gateway. >>> I'm sure a google for "asterisk username authentication" and "freeswitch >>> gateway username" will give you plenty of examples - you will want >>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >>> Google is your friend.. >>> >>> >>> On 2 Feb 2016, at 19:03, Marc S wrote: >>> >>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>> >>> >>> >>> >>> >>> >>> >>> Thanks >>> >>> >>> >>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>> >>>> What is way you planing to use for link DID with user? >>>> >>>> Sergey >>>> >>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>> >>>>> Hello, >>>>> sorry for asking again : >>>>> >>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>> >>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, >>>>> it is the s extension. >>>>> >>>>> Here my bridge tests >>>>> >>>>> >>>>> >>>>> => s extension in asterisk instead of extension >>>>> >>>>> >>>>> >>>>> => Not authenticated in asterisk (because no IP authentication in >>>>> asterisk) >>>>> >>>>> Have you an idea how to send real extension instead of s extension ? >>>>> Thanks >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>> >>>>>> Hello, >>>>>> >>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>> >>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>> >>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>> >>>>>> directory/users does not seem to be the solution because in dialplan, >>>>>> destination DID can't be defined, only user id : >>>>>> >>>>>> >>>>>> >>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>>> not for FS to act as registrar. >>>>>> >>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>> >>>>>> Thanks a lot, >>>>>> Marc >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/9b1279a9/attachment.html From blessendor at gmail.com Tue Feb 2 23:10:25 2016 From: blessendor at gmail.com (Alexandr Usov) Date: Tue, 2 Feb 2016 22:10:25 +0200 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: Post you register and incoming sip peer settings from yor asterisk if you want to got further help. 2016-02-02 22:06 GMT+02:00 Marc S : > Thanks, Asterisk is already registred to FS. > > But when incoming call to FS is bridged to registered Asterisk : > > > > FS send SIP Message to asterisk : > > INVITE s@ > > instead of : > > INVITE 12345678@ > > I would like to get 12345678 in asterisk.. > > > > > 2016-02-02 20:57 GMT+01:00 Alexandr Usov : > >> You must have in Asterisk configs something as in my examples. >> >> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. >> With static IP of both FS and Asterisk you not need to do register from >> asterisk. >> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch >> >> >> ;; type=friend means that this peer can be used for outbound and inbound >> calls, so we don't need to create two peer settings block ( [freeswitch-in] >> and [freeswitch-out] ) >> >> [freeswitch] ;; you maybe want to use here your public DID number, as >> well as in username/fromuser settings >> type=friend >> username=asterisk2fs >> fromuser=asterisk2fs >> fromdomain=ip_of_freeswitch >> host=ip_of_freeswitch >> context=from-freeswitch >> secret=strongpassword >> insecure=port,invite >> qualify=yes >> port=5060 >> >> >> Your context [from-freeswitch] must have an extension, named as >> 'freeswitch' (or your DID number) for incoming calls operations. >> >> >> >> >> 2016-02-02 21:28 GMT+02:00 Marc S : >> >>> It seems that "freeswitch gateway username" return all results about >>> setup username and password on FS to register against external SIP gateway, >>> am i wrong ? >>> >>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >>> >>>> Setup username / password authentication on asterisk and set the >>>> corresponding user & pass in your freeswitch gateway. >>>> I'm sure a google for "asterisk username authentication" and >>>> "freeswitch gateway username" will give you plenty of examples - you will >>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge >>>> gateway". >>>> Google is your friend.. >>>> >>>> >>>> On 2 Feb 2016, at 19:03, Marc S wrote: >>>> >>>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>>> >>>>> What is way you planing to use for link DID with user? >>>>> >>>>> Sergey >>>>> >>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>>> >>>>>> Hello, >>>>>> sorry for asking again : >>>>>> >>>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>>> >>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, >>>>>> it is the s extension. >>>>>> >>>>>> Here my bridge tests >>>>>> >>>>>> >>>>>> >>>>>> => s extension in asterisk instead of extension >>>>>> >>>>>> >>>>>> >>>>>> => Not authenticated in asterisk (because no IP authentication in >>>>>> asterisk) >>>>>> >>>>>> Have you an idea how to send real extension instead of s extension ? >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>>> >>>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>>> >>>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>>> >>>>>>> directory/users does not seem to be the solution because in >>>>>>> dialplan, destination DID can't be defined, only user id : >>>>>>> >>>>>>> >>>>>>> >>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>>>> not for FS to act as registrar. >>>>>>> >>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>>> >>>>>>> Thanks a lot, >>>>>>> Marc >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http:// >>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/f3a3b0b8/attachment-0001.html From sos at sokhapkin.dyndns.org Tue Feb 2 23:13:50 2016 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 02 Feb 2016 15:13:50 -0500 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: <1926695.hkLRVMayM7@sos> Look at REGISTER messages from asterisk. They have header Contact: s@ You get what you ask for :-) On Tuesday 02 February 2016 21:06:01 Marc S wrote: > Thanks, Asterisk is already registred to FS. > > But when incoming call to FS is bridged to registered Asterisk : > > > > FS send SIP Message to asterisk : > > INVITE s@ > > instead of : > > INVITE 12345678@ > > I would like to get 12345678 in asterisk.. > > 2016-02-02 20:57 GMT+01:00 Alexandr Usov : > > You must have in Asterisk configs something as in my examples. > > > > ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. > > With static IP of both FS and Asterisk you not need to do register from > > asterisk. > > register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch > > > > > > ;; type=friend means that this peer can be used for outbound and inbound > > calls, so we don't need to create two peer settings block ( > > [freeswitch-in] > > and [freeswitch-out] ) > > > > [freeswitch] ;; you maybe want to use here your public DID number, as well > > as in username/fromuser settings > > type=friend > > username=asterisk2fs > > fromuser=asterisk2fs > > fromdomain=ip_of_freeswitch > > host=ip_of_freeswitch > > context=from-freeswitch > > secret=strongpassword > > insecure=port,invite > > qualify=yes > > port=5060 > > > > > > Your context [from-freeswitch] must have an extension, named as > > 'freeswitch' (or your DID number) for incoming calls operations. > > > > 2016-02-02 21:28 GMT+02:00 Marc S : > >> It seems that "freeswitch gateway username" return all results about > >> setup username and password on FS to register against external SIP > >> gateway, > >> am i wrong ? > >> > >> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : > >>> Setup username / password authentication on asterisk and set the > >>> corresponding user & pass in your freeswitch gateway. > >>> I'm sure a google for "asterisk username authentication" and "freeswitch > >>> gateway username" will give you plenty of examples - you will want > >>> freeswitch to bridge to gateway - again Google "freeswitch bridge > >>> gateway". > >>> Google is your friend.. > >>> > >>> > >>> On 2 Feb 2016, at 19:03, Marc S wrote: > >>> > >>> I want to use mod xml curl to generate dynamic dialplan xml like this : > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> Thanks > >>> > >>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : > >>>> What is way you planing to use for link DID with user? > >>>> > >>>> Sergey > >>>> > >>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: > >>>>> Hello, > >>>>> sorry for asking again : > >>>>> > >>>>> I hav an asterisk that register on Freeswitch (as a user). > >>>>> > >>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, > >>>>> it is the s extension. > >>>>> > >>>>> Here my bridge tests > >>>>> > >>>>> > >>>>> > >>>>> => s extension in asterisk instead of extension > >>>>> > >>>>> > >>>>> > >>>>> => Not authenticated in asterisk (because no IP authentication in > >>>>> asterisk) > >>>>> > >>>>> Have you an idea how to send real extension instead of s extension ? > >>>>> Thanks > >>>>> > >>>>> 2016-01-03 10:45 GMT+01:00 Marc S : > >>>>>> Hello, > >>>>>> > >>>>>> i'm discovering FS. I hav read a lot about users and gateways. > >>>>>> > >>>>>> I would like to FS act as registrar for authenticated SIP trunking. > >>>>>> > >>>>>> - Customers IPBX would register with login/password to Freeswitch. > >>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. > >>>>>> > >>>>>> directory/users does not seem to be the solution because in dialplan, > >>>>>> destination DID can't be defined, only user id : > >>>>>> > >>>>>> > >>>>>> > >>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, > >>>>>> not for FS to act as registrar. > >>>>>> > >>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? > >>>>>> > >>>>>> Thanks a lot, > >>>>>> Marc > >>>>> > >>>>> ______________________________________________________________________ > >>>>> ___ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>> > >>>> _______________________________________________________________________ > >>>> __ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://confluence.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> ________________________________________________________________________ > >>> _ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> ________________________________________________________________________ > >>> _ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From brian at freeswitch.org Tue Feb 2 23:14:31 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Feb 2016 14:14:31 -0600 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: This is because Asterisk will register as s@ on the contact when it registers to FreeSWITCH, so you can fix this by fixing your register line in sip.conf on Asterisk to register as 12345678 vs the default s If I recall its this format: register => user[:secret[:authuser]]@host[:port][/extension] Thanks, On Tue, Feb 2, 2016 at 2:06 PM, Marc S wrote: > Thanks, Asterisk is already registred to FS. > > But when incoming call to FS is bridged to registered Asterisk : > > > > FS send SIP Message to asterisk : > > INVITE s@ > > instead of : > > INVITE 12345678@ > > I would like to get 12345678 in asterisk.. > > > > > 2016-02-02 20:57 GMT+01:00 Alexandr Usov : > >> You must have in Asterisk configs something as in my examples. >> >> ;;Register on freeswitch if your asterisk behind dynamic IP or NAT, etc. >> With static IP of both FS and Asterisk you not need to do register from >> asterisk. >> register => asterisk2fs:strongpassword at ip_of_freeswitch/freeswitch >> >> >> ;; type=friend means that this peer can be used for outbound and inbound >> calls, so we don't need to create two peer settings block ( [freeswitch-in] >> and [freeswitch-out] ) >> >> [freeswitch] ;; you maybe want to use here your public DID number, as >> well as in username/fromuser settings >> type=friend >> username=asterisk2fs >> fromuser=asterisk2fs >> fromdomain=ip_of_freeswitch >> host=ip_of_freeswitch >> context=from-freeswitch >> secret=strongpassword >> insecure=port,invite >> qualify=yes >> port=5060 >> >> >> Your context [from-freeswitch] must have an extension, named as >> 'freeswitch' (or your DID number) for incoming calls operations. >> >> >> >> >> 2016-02-02 21:28 GMT+02:00 Marc S : >> >>> It seems that "freeswitch gateway username" return all results about >>> setup username and password on FS to register against external SIP gateway, >>> am i wrong ? >>> >>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >>> >>>> Setup username / password authentication on asterisk and set the >>>> corresponding user & pass in your freeswitch gateway. >>>> I'm sure a google for "asterisk username authentication" and >>>> "freeswitch gateway username" will give you plenty of examples - you will >>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge >>>> gateway". >>>> Google is your friend.. >>>> >>>> >>>> On 2 Feb 2016, at 19:03, Marc S wrote: >>>> >>>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>>> >>>>> What is way you planing to use for link DID with user? >>>>> >>>>> Sergey >>>>> >>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>>> >>>>>> Hello, >>>>>> sorry for asking again : >>>>>> >>>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>>> >>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, >>>>>> it is the s extension. >>>>>> >>>>>> Here my bridge tests >>>>>> >>>>>> >>>>>> >>>>>> => s extension in asterisk instead of extension >>>>>> >>>>>> >>>>>> >>>>>> => Not authenticated in asterisk (because no IP authentication in >>>>>> asterisk) >>>>>> >>>>>> Have you an idea how to send real extension instead of s extension ? >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>>> >>>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>>> >>>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>>> >>>>>>> directory/users does not seem to be the solution because in >>>>>>> dialplan, destination DID can't be defined, only user id : >>>>>>> >>>>>>> >>>>>>> >>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>>>> not for FS to act as registrar. >>>>>>> >>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>>> >>>>>>> Thanks a lot, >>>>>>> Marc >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http:// >>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/3b01c323/attachment-0001.html From omortimer at gmail.com Tue Feb 2 23:29:54 2016 From: omortimer at gmail.com (Oz Mortimer) Date: Tue, 2 Feb 2016 20:29:54 +0000 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> Message-ID: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> I think you need to explain what you are trying to achieve.. What you are doing currently is registering a user against freeswitch - not dissimilar to logging in to Skype. In your bridge statement you are calling the registered user - again like you would in Skype. Is there a reason you want asterisk to register against freeswitch? The only thing I can think of is that it's on a private LAN and that your asterisk box does something "special".. > On 2 Feb 2016, at 19:28, Marc S wrote: > > It seems that "freeswitch gateway username" return all results about setup username and password on FS to register against external SIP gateway, am i wrong ? > > 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >> Setup username / password authentication on asterisk and set the corresponding user & pass in your freeswitch gateway. >> I'm sure a google for "asterisk username authentication" and "freeswitch gateway username" will give you plenty of examples - you will want freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >> Google is your friend.. >> >> >>> On 2 Feb 2016, at 19:03, Marc S wrote: >>> >>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>> >>> >>> >>> >>> >>> >>> >>> Thanks >>> >>> >>> >>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>>> What is way you planing to use for link DID with user? >>>> >>>> Sergey >>>> >>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>>> Hello, >>>>> sorry for asking again : >>>>> >>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>> >>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it is the s extension. >>>>> >>>>> Here my bridge tests >>>>> >>>>> >>>>> >>>>> => s extension in asterisk instead of extension >>>>> >>>>> >>>>> >>>>> => Not authenticated in asterisk (because no IP authentication in asterisk) >>>>> >>>>> Have you an idea how to send real extension instead of s extension ? >>>>> Thanks >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>>> Hello, >>>>>> >>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>> >>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>> >>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>> >>>>>> directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id : >>>>>> >>>>>> >>>>>> >>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar. >>>>>> >>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>> >>>>>> Thanks a lot, >>>>>> Marc >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/afa0b676/attachment.html From therebel22 at gmail.com Tue Feb 2 23:42:09 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 21:42:09 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> Message-ID: Thanks for all replies. FS would be used as SBC. I would like my customers' IPBX (Asterisk, Alcatel, ..) register against FS, as a SIP trunk and then i would like routing several DID (not only s) on incoming call to FS to customers IPBX with bridge action. I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, because of security. But registring user in FS seems to expect SIP phone only as user to register, not IPBX as user : Here my asterisk Register : REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport From: ;tag=as47a2d14b To: Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1 CSeq: 103 REGISTER User-Agent: Asterisk Max-Forwards: 70 Authorization: Digest username="testipbx2", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", nc=00000001 Expires: 120 Contact: Event: registration Content-Length: 0 Here FS->Asterisk invite on incoming call INVITE sip:s at 5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c Max-Forwards: 69 From: "45678776" ;tag=4v1S65HQ930mF To: Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 CSeq: 86886190 INVITE Contact: User-Agent: FS/1.6 .. Thanks 2016-02-02 21:29 GMT+01:00 Oz Mortimer : > I think you need to explain what you are trying to achieve.. > What you are doing currently is registering a user against freeswitch - > not dissimilar to logging in to Skype. > In your bridge statement you are calling the registered user - again like > you would in Skype. > Is there a reason you want asterisk to register against freeswitch? The > only thing I can think of is that it's on a private LAN and that your > asterisk box does something "special".. > > > On 2 Feb 2016, at 19:28, Marc S wrote: > > It seems that "freeswitch gateway username" return all results about setup > username and password on FS to register against external SIP gateway, am i > wrong ? > > 2016-02-02 20:23 GMT+01:00 Oz Mortimer : > >> Setup username / password authentication on asterisk and set the >> corresponding user & pass in your freeswitch gateway. >> I'm sure a google for "asterisk username authentication" and "freeswitch >> gateway username" will give you plenty of examples - you will want >> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >> Google is your friend.. >> >> >> On 2 Feb 2016, at 19:03, Marc S wrote: >> >> I want to use mod xml curl to generate dynamic dialplan xml like this : >> >> >> >> >> >> >> >> Thanks >> >> >> >> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >> >>> What is way you planing to use for link DID with user? >>> >>> Sergey >>> >>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>> >>>> Hello, >>>> sorry for asking again : >>>> >>>> I hav an asterisk that register on Freeswitch (as a user). >>>> >>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>>> is the s extension. >>>> >>>> Here my bridge tests >>>> >>>> >>>> >>>> => s extension in asterisk instead of extension >>>> >>>> >>>> >>>> => Not authenticated in asterisk (because no IP authentication in >>>> asterisk) >>>> >>>> Have you an idea how to send real extension instead of s extension ? >>>> Thanks >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>> >>>>> Hello, >>>>> >>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>> >>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>> >>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>> >>>>> directory/users does not seem to be the solution because in dialplan, >>>>> destination DID can't be defined, only user id : >>>>> >>>>> >>>>> >>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>> not for FS to act as registrar. >>>>> >>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>> >>>>> Thanks a lot, >>>>> Marc >>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/c6936c6c/attachment-0001.html From jprangi at gmail.com Tue Feb 2 23:47:50 2016 From: jprangi at gmail.com (Jai Rangi) Date: Tue, 2 Feb 2016 12:47:50 -0800 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> Message-ID: We used LUA script to deal with registered asterisk systems. Example DID 888 555 6666 is assigned to user 101, registered on some ip say 8.8.8.8 Contact will look like s at 8.8.8.8 or 101 at 8.8.8.8.8 LUA Code DIALED_NUMBER=8885556666 USER = 111 location = api:execute("sofia_contact", USER) or "" destinationlocation=location:gsub("sip:(.-)@","sip:"..DIALED_NUMBER.."@") -- This will replace user part in contact heard, (s or even username) with the DID number. -- destinationlocation will look like sip:8885556666 at 8.8.8.8 if (destinationlocation ~= "") then session:execute("bridge","{sip_cid_type=rpid,loop=3}"..destinationlocation) end Hope this help. *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 On Tue, Feb 2, 2016 at 12:29 PM, Oz Mortimer wrote: > I think you need to explain what you are trying to achieve.. > What you are doing currently is registering a user against freeswitch - > not dissimilar to logging in to Skype. > In your bridge statement you are calling the registered user - again like > you would in Skype. > Is there a reason you want asterisk to register against freeswitch? The > only thing I can think of is that it's on a private LAN and that your > asterisk box does something "special".. > > > On 2 Feb 2016, at 19:28, Marc S wrote: > > It seems that "freeswitch gateway username" return all results about setup > username and password on FS to register against external SIP gateway, am i > wrong ? > > 2016-02-02 20:23 GMT+01:00 Oz Mortimer : > >> Setup username / password authentication on asterisk and set the >> corresponding user & pass in your freeswitch gateway. >> I'm sure a google for "asterisk username authentication" and "freeswitch >> gateway username" will give you plenty of examples - you will want >> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >> Google is your friend.. >> >> >> On 2 Feb 2016, at 19:03, Marc S wrote: >> >> I want to use mod xml curl to generate dynamic dialplan xml like this : >> >> >> >> >> >> >> >> Thanks >> >> >> >> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >> >>> What is way you planing to use for link DID with user? >>> >>> Sergey >>> >>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>> >>>> Hello, >>>> sorry for asking again : >>>> >>>> I hav an asterisk that register on Freeswitch (as a user). >>>> >>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>>> is the s extension. >>>> >>>> Here my bridge tests >>>> >>>> >>>> >>>> => s extension in asterisk instead of extension >>>> >>>> >>>> >>>> => Not authenticated in asterisk (because no IP authentication in >>>> asterisk) >>>> >>>> Have you an idea how to send real extension instead of s extension ? >>>> Thanks >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>> >>>>> Hello, >>>>> >>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>> >>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>> >>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>> >>>>> directory/users does not seem to be the solution because in dialplan, >>>>> destination DID can't be defined, only user id : >>>>> >>>>> >>>>> >>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>> not for FS to act as registrar. >>>>> >>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>> >>>>> Thanks a lot, >>>>> Marc >>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/e0fe63b0/attachment.html From brian at freeswitch.org Tue Feb 2 23:50:22 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Feb 2016 14:50:22 -0600 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> Message-ID: You seem like you're trying to fly before you can crawl with FreeSWITCH. You have some base level technological understanding to acquire prior to getting to the end results. You probably didn't read what I wrote in my previous email. Your device is registering as sip:s at 5.6.7.8, so we're gonna follow the rules and call you back at sip:s at 5.6.7.8 You can however in FreeSWITCH force the auth user to match the user and there by having exactly one DID per sip account, or you could change the RURI (which an example of how to do this is in the vanilla config we ship) Here it is: On Tue, Feb 2, 2016 at 2:42 PM, Marc S wrote: > Thanks for all replies. > > FS would be used as SBC. > > I would like my customers' IPBX (Asterisk, Alcatel, ..) register against > FS, as a SIP trunk and then i would like routing several DID (not only s) > on incoming call to FS to customers IPBX with bridge action. > > I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, > because of security. > > But registring user in FS seems to expect SIP phone only as user to > register, not IPBX as user : > > Here my asterisk Register : > > REGISTER sip:1.2.3.4 SIP/2.0 > Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport > From: ;tag=as47a2d14b > To: > Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1 > CSeq: 103 REGISTER > User-Agent: Asterisk > Max-Forwards: 70 > Authorization: Digest username="testipbx2", realm="1.2.3.4", > algorithm=MD5, uri="sip:1.2.3.4", > nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", > response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", > nc=00000001 > Expires: 120 > Contact: > Event: registration > Content-Length: 0 > > Here FS->Asterisk invite on incoming call > > INVITE sip:s at 5.6.7.8 SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c > Max-Forwards: 69 > From: "45678776" ;tag=4v1S65HQ930mF > To: > Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 > CSeq: 86886190 INVITE > Contact: > User-Agent: FS/1.6 > .. > Thanks > > > > > > 2016-02-02 21:29 GMT+01:00 Oz Mortimer : > >> I think you need to explain what you are trying to achieve.. >> What you are doing currently is registering a user against freeswitch - >> not dissimilar to logging in to Skype. >> In your bridge statement you are calling the registered user - again like >> you would in Skype. >> Is there a reason you want asterisk to register against freeswitch? The >> only thing I can think of is that it's on a private LAN and that your >> asterisk box does something "special".. >> >> >> On 2 Feb 2016, at 19:28, Marc S wrote: >> >> It seems that "freeswitch gateway username" return all results about >> setup username and password on FS to register against external SIP gateway, >> am i wrong ? >> >> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >> >>> Setup username / password authentication on asterisk and set the >>> corresponding user & pass in your freeswitch gateway. >>> I'm sure a google for "asterisk username authentication" and "freeswitch >>> gateway username" will give you plenty of examples - you will want >>> freeswitch to bridge to gateway - again Google "freeswitch bridge gateway". >>> Google is your friend.. >>> >>> >>> On 2 Feb 2016, at 19:03, Marc S wrote: >>> >>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>> >>> >>> >>> >>> >>> >>> >>> Thanks >>> >>> >>> >>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>> >>>> What is way you planing to use for link DID with user? >>>> >>>> Sergey >>>> >>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>> >>>>> Hello, >>>>> sorry for asking again : >>>>> >>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>> >>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, >>>>> it is the s extension. >>>>> >>>>> Here my bridge tests >>>>> >>>>> >>>>> >>>>> => s extension in asterisk instead of extension >>>>> >>>>> >>>>> >>>>> => Not authenticated in asterisk (because no IP authentication in >>>>> asterisk) >>>>> >>>>> Have you an idea how to send real extension instead of s extension ? >>>>> Thanks >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>> >>>>>> Hello, >>>>>> >>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>> >>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>> >>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>> >>>>>> directory/users does not seem to be the solution because in dialplan, >>>>>> destination DID can't be defined, only user id : >>>>>> >>>>>> >>>>>> >>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>>> not for FS to act as registrar. >>>>>> >>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>> >>>>>> Thanks a lot, >>>>>> Marc >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/fffdc3f7/attachment-0001.html From therebel22 at gmail.com Wed Feb 3 00:00:10 2016 From: therebel22 at gmail.com (Marc S) Date: Tue, 2 Feb 2016 22:00:10 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: <490EAD37-3FE1-4A94-8166-499B461F9C53@gmail.com> <1CB79FBF-A771-4A02-AD43-70597741BA9D@gmail.com> Message-ID: Thanks a lot for reply ! Sorry for misunderstanding. I'm going to study what you wrote. 2016-02-02 21:50 GMT+01:00 Brian West : > You seem like you're trying to fly before you can crawl with FreeSWITCH. > You have some base level technological understanding to acquire prior to > getting to the end results. You probably didn't read what I wrote in my > previous email. Your device is registering as sip:s at 5.6.7.8, so we're > gonna follow the rules and call you back at sip:s at 5.6.7.8 > > You can however in FreeSWITCH force the auth user to match the user and > there by having exactly one DID per sip account, or you could change the > RURI (which an example of how to do this is in the vanilla config we ship) > > Here it is: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Tue, Feb 2, 2016 at 2:42 PM, Marc S wrote: > >> Thanks for all replies. >> >> FS would be used as SBC. >> >> I would like my customers' IPBX (Asterisk, Alcatel, ..) register against >> FS, as a SIP trunk and then i would like routing several DID (not only s) >> on incoming call to FS to customers IPBX with bridge action. >> >> I prefer IPBX login/pass auth against FS rather than IPBX IP auth only, >> because of security. >> >> But registring user in FS seems to expect SIP phone only as user to >> register, not IPBX as user : >> >> Here my asterisk Register : >> >> REGISTER sip:1.2.3.4 SIP/2.0 >> Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK45872adf;rport >> From: ;tag=as47a2d14b >> To: >> Call-ID: 6896d4fe6ab4904b7830df881a25e4cf at 127.0.1.1 >> CSeq: 103 REGISTER >> User-Agent: Asterisk >> Max-Forwards: 70 >> Authorization: Digest username="testipbx2", realm="1.2.3.4", >> algorithm=MD5, uri="sip:1.2.3.4", >> nonce="618f279c-c9ea-11e5-9551-4749c6ba93f1", >> response="332db701f8c8404f4f1624b9f6c68f2c", qop=auth, cnonce="751ce1f8", >> nc=00000001 >> Expires: 120 >> Contact: >> Event: registration >> Content-Length: 0 >> >> Here FS->Asterisk invite on incoming call >> >> INVITE sip:s at 5.6.7.8 SIP/2.0 >> Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgaUt932FD952c >> Max-Forwards: 69 >> From: "45678776" ;tag=4v1S65HQ930mF >> To: >> Call-ID: 971c11f4-c9ea-11e5-956b-4749c6ba93f1 >> CSeq: 86886190 INVITE >> Contact: >> User-Agent: FS/1.6 >> .. >> Thanks >> >> >> >> >> >> 2016-02-02 21:29 GMT+01:00 Oz Mortimer : >> >>> I think you need to explain what you are trying to achieve.. >>> What you are doing currently is registering a user against freeswitch - >>> not dissimilar to logging in to Skype. >>> In your bridge statement you are calling the registered user - again >>> like you would in Skype. >>> Is there a reason you want asterisk to register against freeswitch? The >>> only thing I can think of is that it's on a private LAN and that your >>> asterisk box does something "special".. >>> >>> >>> On 2 Feb 2016, at 19:28, Marc S wrote: >>> >>> It seems that "freeswitch gateway username" return all results about >>> setup username and password on FS to register against external SIP gateway, >>> am i wrong ? >>> >>> 2016-02-02 20:23 GMT+01:00 Oz Mortimer : >>> >>>> Setup username / password authentication on asterisk and set the >>>> corresponding user & pass in your freeswitch gateway. >>>> I'm sure a google for "asterisk username authentication" and >>>> "freeswitch gateway username" will give you plenty of examples - you will >>>> want freeswitch to bridge to gateway - again Google "freeswitch bridge >>>> gateway". >>>> Google is your friend.. >>>> >>>> >>>> On 2 Feb 2016, at 19:03, Marc S wrote: >>>> >>>> I want to use mod xml curl to generate dynamic dialplan xml like this : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >>>> >>>>> What is way you planing to use for link DID with user? >>>>> >>>>> Sergey >>>>> >>>>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>>>> >>>>>> Hello, >>>>>> sorry for asking again : >>>>>> >>>>>> I hav an asterisk that register on Freeswitch (as a user). >>>>>> >>>>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, >>>>>> it is the s extension. >>>>>> >>>>>> Here my bridge tests >>>>>> >>>>>> >>>>>> >>>>>> => s extension in asterisk instead of extension >>>>>> >>>>>> >>>>>> >>>>>> => Not authenticated in asterisk (because no IP authentication in >>>>>> asterisk) >>>>>> >>>>>> Have you an idea how to send real extension instead of s extension ? >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>>>> >>>>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>>>> >>>>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>>>> >>>>>>> directory/users does not seem to be the solution because in >>>>>>> dialplan, destination DID can't be defined, only user id : >>>>>>> >>>>>>> >>>>>>> >>>>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>>>> not for FS to act as registrar. >>>>>>> >>>>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>>>> >>>>>>> Thanks a lot, >>>>>>> Marc >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http:// >>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/f77b49f5/attachment-0001.html From govoiper at gmail.com Wed Feb 3 03:44:09 2016 From: govoiper at gmail.com (SamyGo) Date: Tue, 2 Feb 2016 19:44:09 -0500 Subject: [Freeswitch-users] Storm of NOTIFY from FS Message-ID: Hi All, Ive few FS servers which solely handle Parking lots and their states behind Kamailio SBC. It has all been running very good for about two years now. Just as of this morning all of those FS servers have been sending storm of NOTIFY messages to Kamailio servers. This in turn consumes all CPU of FS server and fs cli becomes just irresponsive. I'm investigating whether Kamailio have blocked FS servers via pike module and hence FS servers go crazy on resending NOTIFY ever more aggressively. So far it doesnt look like it. I just want to ask under what circumstances that even after restarting service FS it would pick on all the states and resume sending to SBC. Is there a cache/DB somehwere which causes new FS process to continue with NOTIFY storm ? Ive FS version 1.5 deployed. Thanks in advance for any help. Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160202/66cc8064/attachment.html From nandy1925 at gmail.com Wed Feb 3 09:16:57 2016 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 3 Feb 2016 06:16:57 +0000 Subject: [Freeswitch-users] FreeSWITCH vs Asterisk on Reddit... In-Reply-To: References: <51f601d137c9$ae032190$0a0964b0$@freeswitch.org> Message-ID: My personal experience. I rescued a client by replacing his Elastix with FS because it's dropping calls. On Wed, Dec 16, 2015 at 8:57 PM, Larry Morley wrote: > Regarding this portion of Ken's comment in particular: > > "... people think that before there is Free in the name or that it is open > source, that the developers of the project are doing this for free. ..." > > That attitude by no means applies to open source software alone; it's been > my experience that you'll most certainly find the same attitudes and > erroneous beliefs in play, in the realms of free works in general - e.g., > free or public domain hardware, software, artwork, music, writing, circuit > designs. Those attitudes and beliefs are the reason why it's so uncommon > for companies to offer free phone based technical support anymore. They're > the reason why when you do find phone based tech support, or, for example, > report a trouble to a phone company, cable company, ITSP, etc., the > response you get is tiered. In the former, they know there's a good chance > that the real reason for the call is because the caller didn't want to > bother reading the documentation supplied with the product. Which, over > time, has resulted in less documentation being supplied with many products > - I've witnessed first hand "the bean counters" opting to not spend what > they would have in years past to hire technical writers, choosing instead > to have an intern or recent hire produce something the company can put in > the box. > > I firmly believe these attitudes stem from a sense of entitlement. And a > belief on the part of some that their time, their problem, their existence, > is more important than anyone else's. > > Fortunately, I also know that history is a cyclical beast, and that at > some point, the attitudes and fundamental beliefs of perhaps not everyone, > but at least of the greater society, will likely return, for a while at any > rate - for history is cyclical - to a point where, bearing in mind that > only a healthy, whole, grounded person has the ability to give of > themselves let alone anything with giving - people will tend to be far more > concerned with what they can contribute to others, both now and for > posterity - and will choose to act in accordance with their beliefs of > their own accord, their own free wills - than with what they themselves can > accumulate and with what others can do for them. > > I welcome the arrival of that day. > > Larry Morley > > On Dec 16, 2015 01:20, "Ken Rice" wrote: > > > > That guy got cranky because he thought that the developers time was > free? For some reason people think that before there is Free in the name or > that it is open source, that the developers of the project are doing this > for free. We all have families to feed. Unlike some open source projects, > FreeSWITCH is not funded by millions of dollars of funding, every bit of > funding comes from people using FreeSWITCH and contracting the FreeSWITCH > Team to help them deploy, configure, or enhance the software. I can?t think > of anyone of the supporters that has wanted to keep any code additions out > of tree. > > > > > > > > So even if you don?t have a huge budget you can still help out Anthony, > Brian, Mike, Me, and William. Contact us, let us help with your project. > Sure, there may be a charge for that initial consultation, but it also > allows us to dedicate time with you to engineer a plan to help you reach > your goals. > > > > > > > > Want to send a little Christmas or Hanukkah gift to one of the > Developers? Visit https://freeswitch.org/core-team/, Our wishlists are > there. > > > > Want to but the dev?s dinner or something Similar? There?s a donate > button right on the website. > > > > > > > > Want to help in other ways? > > > > Join the Docs Team and help document things on Confluence. We still have > a several hundred pages on the old wiki that need to be clean up, updated > and migrated to Confluence. > > > > Help us sort thru the bugs and test patches and pull requests. Anyone > can comment on open tickets and add information? Want to be an official bug > marshall? Email me (krice at freeswitch.org) or brian at freeswitch.org off > list and we?ll help get you started. > > > > Join the FreeSWITCH Team via HipChat at https://hipchat.freeswitch.org/ > hang out meet the devs and chat with other FreeSWITCH users. > > > > Like IRC? Check out #FreeSWITCH on freenode.net > > > > > > > > Every Little Bit Helps! > > > > > > > > K > > > > > > > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of chris > > Sent: Tuesday, December 15, 2015 11:38 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FreeSWITCH vs Asterisk on Reddit... > > > > > > > > about on par with this one which was interesting: > > > > > > > > > https://www.reddit.com/r/freeswitch/comments/3o5acr/i_failed_with_freeswitch/ > > > > > > > > On Tue, Dec 15, 2015 at 6:12 PM, Brian West > wrote: > >> > >> Interesting thread... > >> > >> > >> > >> https://www.reddit.com/r/VOIP/comments/3wy9h8/freeswitch_vs_asterisk/ > >> > >> > >> > >> Everyone should check it out. > >> > >> > >> > >> Thanks, > >> > >> > >> > >> > >> -- > >> > >> Brian West > >> brian at freeswitch.org > >> > >> Twitter: @FreeSWITCH , @briankwest > >> http://www.freeswitchbook.com > >> http://www.freeswitchcookbook.com > >> > >> Got Bugs? Report them here! | Reddit: /r/freeswitch > >> > >> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > >> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/a924affe/attachment.html From s.safarov at gmail.com Wed Feb 3 09:42:21 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 3 Feb 2016 09:42:21 +0300 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: Read theread "Conectin a SPA3102 as pstn gateway on FS " {sip_invite_to_uri=}user/ reg_user at mydomain.org If requred mutal autertication {sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri=}user/ reg_user at mydomain.org On Tue, Feb 2, 2016 at 10:03 PM, Marc S wrote: > I want to use mod xml curl to generate dynamic dialplan xml like this : > > > > > > > > Thanks > > > > 2016-02-02 19:16 GMT+01:00 Sergey Safarov : > >> What is way you planing to use for link DID with user? >> >> Sergey >> >> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >> >>> Hello, >>> sorry for asking again : >>> >>> I hav an asterisk that register on Freeswitch (as a user). >>> >>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>> is the s extension. >>> >>> Here my bridge tests >>> >>> >>> >>> => s extension in asterisk instead of extension >>> >>> >>> >>> => Not authenticated in asterisk (because no IP authentication in >>> asterisk) >>> >>> Have you an idea how to send real extension instead of s extension ? >>> Thanks >>> >>> >>> >>> >>> >>> >>> >>> 2016-01-03 10:45 GMT+01:00 Marc S : >>> >>>> Hello, >>>> >>>> i'm discovering FS. I hav read a lot about users and gateways. >>>> >>>> I would like to FS act as registrar for authenticated SIP trunking. >>>> >>>> - Customers IPBX would register with login/password to Freeswitch. >>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>> >>>> directory/users does not seem to be the solution because in dialplan, >>>> destination DID can't be defined, only user id : >>>> >>>> >>>> >>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>> not for FS to act as registrar. >>>> >>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>> >>>> Thanks a lot, >>>> Marc >>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/3cc712ef/attachment-0001.html From therebel22 at gmail.com Wed Feb 3 11:25:18 2016 From: therebel22 at gmail.com (Marc S) Date: Wed, 3 Feb 2016 09:25:18 +0100 Subject: [Freeswitch-users] Freeswitch as Registrar for SIP Trunk In-Reply-To: References: Message-ID: I will try soon, thanks 2016-02-03 7:42 GMT+01:00 Sergey Safarov : > Read theread "Conectin a SPA3102 as pstn gateway on FS > > " > > {sip_invite_to_uri=}user/ > reg_user at mydomain.org > > If requred mutal autertication > > > {sip_auth_username='login_name',sip_auth_password='strong_password_here',sip_invite_to_uri= destination_number}@mydomain.org>}user/ reg_user at mydomain.org > > > On Tue, Feb 2, 2016 at 10:03 PM, Marc S wrote: > >> I want to use mod xml curl to generate dynamic dialplan xml like this : >> >> >> >> >> >> >> >> Thanks >> >> >> >> 2016-02-02 19:16 GMT+01:00 Sergey Safarov : >> >>> What is way you planing to use for link DID with user? >>> >>> Sergey >>> >>> On Tue, Feb 2, 2016 at 8:20 PM, Marc S wrote: >>> >>>> Hello, >>>> sorry for asking again : >>>> >>>> I hav an asterisk that register on Freeswitch (as a user). >>>> >>>> When a call is incoming to FS, FS send it to asterisk : In asterisk, it >>>> is the s extension. >>>> >>>> Here my bridge tests >>>> >>>> >>>> >>>> => s extension in asterisk instead of extension >>>> >>>> >>>> >>>> => Not authenticated in asterisk (because no IP authentication in >>>> asterisk) >>>> >>>> Have you an idea how to send real extension instead of s extension ? >>>> Thanks >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2016-01-03 10:45 GMT+01:00 Marc S : >>>> >>>>> Hello, >>>>> >>>>> i'm discovering FS. I hav read a lot about users and gateways. >>>>> >>>>> I would like to FS act as registrar for authenticated SIP trunking. >>>>> >>>>> - Customers IPBX would register with login/password to Freeswitch. >>>>> - Incoming call would be routed to these SIP trunks in dialplan XML. >>>>> >>>>> directory/users does not seem to be the solution because in dialplan, >>>>> destination DID can't be defined, only user id : >>>>> >>>>> >>>>> >>>>> gateway seems to be designed for SIP trunking to remote SIP gateway, >>>>> not for FS to act as registrar. >>>>> >>>>> Is it possible to FS to act as authenticated SIP trunking registrar ? >>>>> >>>>> Thanks a lot, >>>>> Marc >>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/4ce21f9f/attachment.html From yadenis at seznam.cz Wed Feb 3 11:20:11 2016 From: yadenis at seznam.cz (Denis Jakovlev) Date: Wed, 3 Feb 2016 09:20:11 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> Message-ID: <628620366.20160203092011@seznam.cz> Dobr? den, The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment) My clients use different browsers and I can not force them to use something specific. With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste: With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. On 1 February 2016 at 17:12, Ken Rice wrote: Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/cb18950e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Capture20.JPG Type: image/jpeg Size: 36404 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/cb18950e/attachment-0001.jpe From krice at freeswitch.org Wed Feb 3 13:03:29 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 3 Feb 2016 04:03:29 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <628620366.20160203092011@seznam.cz> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> <628620366.20160203092011@seznam.cz> Message-ID: <590d01d15e6a$22651810$672f4830$@freeswitch.org> Oh So where is the Jira on this it doesn?t work in firefox with the debugging information? If you know of an issue like this you should report it to jira (https://freeswitch.org/jira) so a dev can try to replicate and fix it? You cant expect bugs to get fixed if you aren?t reporting them properly? Go troll somewhere else?. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Wednesday, February 3, 2016 2:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment) My clients use different browsers and I can not force them to use something specific. With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste: With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. On 1 February 2016 at 17:12, Ken Rice < krice at freeswitch.org> wrote: Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org> Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/a2c5aaef/attachment.html From mike at jerris.com Wed Feb 3 19:18:53 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 3 Feb 2016 11:18:53 -0500 Subject: [Freeswitch-users] mod_potaudio eror on opening device In-Reply-To: References: Message-ID: <3B5F650F-5704-40CC-AA6D-5414AF05BBE4@jerris.com> I know that it is available in the source code, I do not know if packages install these or not, I suspect not. > On Jan 30, 2016, at 6:56 AM, Pete Kay wrote: > > Hi Michael > > Thanks alot for your reply. > > portaudio was installed as part of freeswitch. > > I search and could not find the test program. May I know where I can find this test program to test? > > P > > On Mon, Jan 25, 2016 at 4:05 AM, Michael Jerris > wrote: > if you build port audio manually do the sample programs work? > > > On Sunday, January 24, 2016, Pete Kay > wrote: > Hi > <> > > I am not able to get pa play to work. could someone please kindly help me out? I am getting the following ERROR messages: > > > <> > freeswitch at vps57327.vps.ovh.ca <>> pa play /tmp/running.wav > > Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 > > Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869 > > Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 > > 2016-01-24 07:02:32.591002 [ERR] mod_portaudio.c:2428 Error opening audio device retrying > > Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 > > Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869 > > Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 > > > > Failed to engage audio device > > > > 2016-01-24 07:02:34.090957 [ERR] mod_portaudio.c:2435 Can't open audio device > > freeswitch at vps57327.vps.ovh.ca <>> pa outdev #4 > > > > outdev set to 4 > > > > freeswitch at vps57327.vps.ovh.ca <>> pa indev #5 > > > > indev set to 5 > > > > freeswitch at vps57327.vps.ovh.ca <>> pa play /tmp/running.wav > > Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 > > Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872 > > Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 > > 2016-01-24 07:02:51.991013 [ERR] mod_portaudio.c:2428 Error opening audio device retrying > > Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 > > Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872 > > Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 > > > > Failed to engage audio device > > > > 2016-01-24 07:02:53.491018 [ERR] mod_portaudio.c:2435 Can't open audio device > > freeswitch at vps57327.vps.ovh.ca <>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/70102337/attachment-0001.html From dcolombo at voismart.it Wed Feb 3 19:39:51 2016 From: dcolombo at voismart.it (Davide Colombo) Date: Wed, 3 Feb 2016 17:39:51 +0100 (CET) Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <590d01d15e6a$22651810$672f4830$@freeswitch.org> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> <628620366.20160203092011@seznam.cz> <590d01d15e6a$22651810$672f4830$@freeswitch.org> Message-ID: <1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it> I reported this bug to jira: https://freeswitch.org/jira/browse/FS-8805 ----- Messaggio originale ----- Da: "Ken Rice" A: "freeswitch-users" Inviato: Mercoled?, 3 febbraio 2016 11:03:29 Oggetto: Re: [Freeswitch-users] Verto vs. SIP.js Re: [Freeswitch-users] Verto vs. SIP.js Oh So where is the Jira on this it doesn?t work in firefox with the debugging information? If you know of an issue like this you should report it to jira ( https://freeswitch.org/jira ) so a dev can try to replicate and fix it? You cant expect bugs to get fixed if you aren?t reporting them properly? Go troll somewhere else?. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Wednesday, February 3, 2016 2:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment) My clients use different browsers and I can not force them to use something specific. With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste: With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. On 1 February 2016 at 17:12, Ken Rice < krice at freeswitch.org > wrote: Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel . 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel . 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Wed Feb 3 19:45:21 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 3 Feb 2016 11:45:21 -0500 Subject: [Freeswitch-users] BLF for Gateway In-Reply-To: References: Message-ID: You might be able to do something here with display updates. > On Feb 2, 2016, at 6:58 AM, Roman Dissauer wrote: > > Hi All, > > is there a way to get gateway usage in freeswitch on my phones blf? > I have multiple gateways registered and want to see which one is taken for a particular outbound call. > > Best Regards, > Roman From vfclists at gmail.com Wed Feb 3 20:11:10 2016 From: vfclists at gmail.com (vfclists .) Date: Wed, 3 Feb 2016 17:11:10 +0000 Subject: [Freeswitch-users] How do you access caller profile fields in the XML dialplan? Message-ID: What is the syntax for accessing caller profile fields in the dialplan, for instance if you want to set a channel variable to a caller profile field? Is there also a way of performing some string manipulation and extraction on the variables in the XML dialplan? This is what I am trying to achieve. The CDR from a service provider contains the CLI from a gateway on the customers premises, but it doesn't show which extension on the customer's premises the call came from. What I need is to be able to obtain the extension of the sip device and combine that with the customer gateway's CLI so that it shows in the CDR record. eg if the gateway's CLI is 2340 and the extension of the caller is 1001, corresponding to the UserId on a sipura, I want the CLI passed to the service provider to be 23401001. When I check the XML cdr in Freeswitch I see an XML value which in XPath would be accessed as /callflow/caller_profile/caller_id_number. All the information is therefore passed onto Freeswitch for use in the call, and what I need is to be able to access it and change the effective_caller_id_number before bridging the call. -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/a0852acf/attachment.html From italo at freeswitch.org Wed Feb 3 20:34:58 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 3 Feb 2016 14:34:58 -0300 Subject: [Freeswitch-users] How do you access caller profile fields in the XML dialplan? In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables On Wed, Feb 3, 2016 at 2:11 PM, vfclists . wrote: > What is the syntax for accessing caller profile fields in the dialplan, > for instance if you want to set a channel variable to a caller profile > field? > > Is there also a way of performing some string manipulation and extraction > on the variables in the XML dialplan? > > This is what I am trying to achieve. The CDR from a service provider > contains the CLI from a gateway on the customers premises, but it doesn't > show which extension on the customer's premises the call came from. What I > need is to be able to obtain the extension of the sip device and combine > that with the customer gateway's CLI so that it shows in the CDR record. > > eg if the gateway's CLI is 2340 and the extension of the caller is 1001, > corresponding to the UserId on a sipura, I want the CLI passed to the > service provider to be 23401001. When I check the XML cdr in Freeswitch I > see an XML value which in XPath would be accessed as > /callflow/caller_profile/caller_id_number. All the information is therefore > passed onto Freeswitch for use in the call, and what I need is to be able > to access it and change the effective_caller_id_number before bridging the > call. > > -- > Frank Church > > ======================= > http://devblog.brahmancreations.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/bb6885fd/attachment.html From vfclists at gmail.com Wed Feb 3 21:15:44 2016 From: vfclists at gmail.com (vfclists .) Date: Wed, 3 Feb 2016 18:15:44 +0000 Subject: [Freeswitch-users] How do you access caller profile fields in the XML dialplan? In-Reply-To: References: Message-ID: There are quite a number of variables there, but I can't seem to find tthe one I want because the actual extension doesn't seem to be used in the configuration files. eg. When I check the logs I see output like this sofia.c:1192 sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012> What I want to do is to combine the 2048 with the 1012 to create the effective_caller_id_number 20481012. The information is present somewhere in Freeswitch. The extensions connect to Asterisk because most of the applications were written for Asterisk, but Asterisk proved to be terrible behind NAT so a Freeswitch system was added as an intermediate gateway, so none of the extensions register to the Freeswitch, only the Asterisk. On 3 February 2016 at 17:34, ?talo Rossi wrote: > > https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables > > On Wed, Feb 3, 2016 at 2:11 PM, vfclists . wrote: > >> What is the syntax for accessing caller profile fields in the dialplan, >> for instance if you want to set a channel variable to a caller profile >> field? >> >> Is there also a way of performing some string manipulation and extraction >> on the variables in the XML dialplan? >> >> This is what I am trying to achieve. The CDR from a service provider >> contains the CLI from a gateway on the customers premises, but it doesn't >> show which extension on the customer's premises the call came from. What I >> need is to be able to obtain the extension of the sip device and combine >> that with the customer gateway's CLI so that it shows in the CDR record. >> >> eg if the gateway's CLI is 2340 and the extension of the caller is 1001, >> corresponding to the UserId on a sipura, I want the CLI passed to the >> service provider to be 23401001. When I check the XML cdr in Freeswitch I >> see an XML value which in XPath would be accessed as >> /callflow/caller_profile/caller_id_number. All the information is therefore >> passed onto Freeswitch for use in the call, and what I need is to be able >> to access it and change the effective_caller_id_number before bridging the >> call. >> >> -- >> Frank Church >> >> ======================= >> http://devblog.brahmancreations.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ?talo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/1b19bb42/attachment-0001.html From bobjectsfreeswitch at gmail.com Wed Feb 3 22:28:42 2016 From: bobjectsfreeswitch at gmail.com (Bob Hartwig) Date: Wed, 3 Feb 2016 13:28:42 -0600 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? Message-ID: I have a client that needs to reliably detect if their outbound calls are answered by a human or voicemail system, so that they can take different actions based on that determination. I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. Am I correct about that? How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? Thanks! Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/07d92f80/attachment.html From anthony.minessale at gmail.com Wed Feb 3 22:45:39 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Feb 2016 13:45:39 -0600 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? In-Reply-To: References: Message-ID: There is actually a commercial module for AMD offered by FreeSWITCH Solutions https://freeswitch.com/cart.php?gid=2 On Wed, Feb 3, 2016 at 1:28 PM, Bob Hartwig wrote: > I have a client that needs to reliably detect if their outbound calls are > answered by a human or voicemail system, so that they can take different > actions based on that determination. > > I looked at the AVMD module documentation at > https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it > seems to indicate that this simply detects a beep, i.e. it does not use > talking / silence heuristics into account to determine if the call is > answered by a human or machine. > > Am I correct about that? How is this intended to be used, do I assume > that the call is answered by a human until this module sends > a avmd%3A%3Abeep event to me? How do others use this module or other > techniques to determine human / machine answer to outbound calls with > Freeswitch? > > Thanks! > Bob > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/c754aef5/attachment.html From lists at telefaks.de Wed Feb 3 22:46:38 2016 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 03 Feb 2016 20:46:38 +0100 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? In-Reply-To: References: Message-ID: <56B2591E.1030404@telefaks.de> I have had pretty good results with Sangoma Lyra AMD. Peter On 02/03/16 20:28, Bob Hartwig wrote: > I have a client that needs to reliably detect if their outbound calls > are answered by a human or voicemail system, so that they can take > different actions based on that determination. > > I looked at the AVMD module documentation > at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and > it seems to indicate that this simply detects a beep, i.e. it does not > use talking / silence heuristics into account to determine if the call > is answered by a human or machine. > > Am I correct about that? How is this intended to be used, do I assume > that the call is answered by a human until this module sends > a avmd%3A%3Abeep event to me? How do others use this module or other > techniques to determine human / machine answer to outbound calls with > Freeswitch? > > Thanks! > Bob > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/2a6c403d/attachment.html From cmrienzo at gmail.com Wed Feb 3 22:50:58 2016 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 3 Feb 2016 14:50:58 -0500 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? In-Reply-To: References: Message-ID: I wouldn't use beep detection in a dialer application, but it could be useful in something like follow me to reduce the occurrence of voicemails being left on subscriber phones. On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig wrote: > I have a client that needs to reliably detect if their outbound calls are > answered by a human or voicemail system, so that they can take different > actions based on that determination. > > I looked at the AVMD module documentation at > https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it > seems to indicate that this simply detects a beep, i.e. it does not use > talking / silence heuristics into account to determine if the call is > answered by a human or machine. > > Am I correct about that? How is this intended to be used, do I assume > that the call is answered by a human until this module sends > a avmd%3A%3Abeep event to me? How do others use this module or other > techniques to determine human / machine answer to outbound calls with > Freeswitch? > > Thanks! > Bob > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/3acd5fa9/attachment.html From ssinyagin at gmail.com Wed Feb 3 22:57:44 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 3 Feb 2016 20:57:44 +0100 Subject: [Freeswitch-users] How do you access caller profile fields in the XML dialplan? In-Reply-To: References: Message-ID: You need user_data. See the example here: https://txlab.wordpress.com/2013/06/29/freeswitch-limiting-the-number-of-concurrent-calls-on-multiple-sip-accounts/ On 3 Feb 2016 19:16, "vfclists ." wrote: > There are quite a number of variables there, but I can't seem to find tthe > one I want because the actual extension doesn't seem to be used in the > configuration files. > > eg. When I check the logs I see output like this sofia.c:1192 > sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012> > > What I want to do is to combine the 2048 with the 1012 to create the > effective_caller_id_number 20481012. The information is present somewhere > in Freeswitch. The extensions connect to Asterisk because most of the > applications were written for Asterisk, but Asterisk proved to be terrible > behind NAT so a Freeswitch system was added as an intermediate gateway, so > none of the extensions register to the Freeswitch, only the Asterisk. > > > On 3 February 2016 at 17:34, ?talo Rossi wrote: > >> >> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables >> >> On Wed, Feb 3, 2016 at 2:11 PM, vfclists . wrote: >> >>> What is the syntax for accessing caller profile fields in the dialplan, >>> for instance if you want to set a channel variable to a caller profile >>> field? >>> >>> Is there also a way of performing some string manipulation and >>> extraction on the variables in the XML dialplan? >>> >>> This is what I am trying to achieve. The CDR from a service provider >>> contains the CLI from a gateway on the customers premises, but it doesn't >>> show which extension on the customer's premises the call came from. What I >>> need is to be able to obtain the extension of the sip device and combine >>> that with the customer gateway's CLI so that it shows in the CDR record. >>> >>> eg if the gateway's CLI is 2340 and the extension of the caller is 1001, >>> corresponding to the UserId on a sipura, I want the CLI passed to the >>> service provider to be 23401001. When I check the XML cdr in Freeswitch I >>> see an XML value which in XPath would be accessed as >>> /callflow/caller_profile/caller_id_number. All the information is therefore >>> passed onto Freeswitch for use in the call, and what I need is to be able >>> to access it and change the effective_caller_id_number before bridging the >>> call. >>> >>> -- >>> Frank Church >>> >>> ======================= >>> http://devblog.brahmancreations.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> ?talo Rossi >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Frank Church > > ======================= > http://devblog.brahmancreations.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/3fb4a0c1/attachment-0001.html From victor.chukalovskiy at gmail.com Wed Feb 3 23:07:29 2016 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 3 Feb 2016 15:07:29 -0500 Subject: [Freeswitch-users] Is there a way to deal with "false ring-back than 503" carriers? Message-ID: <56B25E01.6030608@gmail.com> Hello, A downstream carrier would accept INVITE, send 183 w/SDP, and almost instantaneously respond "503". This kills the call not letting re-route to the next carrier. Without calling any specific names, this seems to become more common. Unfortunately, often the offending carrier is a few steps down the call path and I can't reach them with a cluebat. Is there a way to make FS not fail the call and try the next route if using LCR as per below? Note that we are in bypass_media mode: Or to keep it simple, forget LCR for a moment, can we make it go to gateway_2 in the following example if gw1 returns "183" followed by "503"? Many thanks, -Victor From vfclists at gmail.com Thu Feb 4 00:11:29 2016 From: vfclists at gmail.com (vfclists .) Date: Wed, 3 Feb 2016 21:11:29 +0000 Subject: [Freeswitch-users] How do you access caller profile fields in the XML dialplan? In-Reply-To: References: Message-ID: This is where the problem is. The place where I need that information is on the external gateway itself, not on the internal one (behind NAT) where the devices register. The directory settings on the internal gateway has the effective_caller_id_name and number changed from their defaults, so the SIP user name and user id on the sip phone don't seem to appear in any of the channel variables during the call. To give more detail. All SIP phone registers to Asterisk user id and password. Asterisk registers to internal Freeswitch running on a different port on the same box. The internal Freeswitch connects to external Freeswitch which connects external providers gateways. I need to combine the internal Gateway's CLI with the User ID from Sipura in the external Freeswitch's to form the effective_caller_id_number to differentiate calls in the external providers CDR. The username and user id on the Sipura are present somewhere but they don't seem to be available in any of the caller profile fields or variables in the A Leg phase. It is in the B Leg that somehow they are obtained, and they appear in info command, and in the Bleg xml_cdr. Can user_data actually retrieve them in the ALeg stage? On 3 February 2016 at 19:57, Stanislav Sinyagin wrote: > You need user_data. > See the example here: > https://txlab.wordpress.com/2013/06/29/freeswitch-limiting-the-number-of-concurrent-calls-on-multiple-sip-accounts/ > On 3 Feb 2016 19:16, "vfclists ." wrote: > >> There are quite a number of variables there, but I can't seem to find >> tthe one I want because the actual extension doesn't seem to be used in the >> configuration files. >> >> eg. When I check the logs I see output like this sofia.c:1192 >> sofia/internal/2048 at 201.182.29.148 Update Caller ID to "Desk 12" <1012> >> >> What I want to do is to combine the 2048 with the 1012 to create the >> effective_caller_id_number 20481012. The information is present somewhere >> in Freeswitch. The extensions connect to Asterisk because most of the >> applications were written for Asterisk, but Asterisk proved to be terrible >> behind NAT so a Freeswitch system was added as an intermediate gateway, so >> none of the extensions register to the Freeswitch, only the Asterisk. >> >> >> On 3 February 2016 at 17:34, ?talo Rossi wrote: >> >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-CallerProfileFieldsvs.ChannelVariables >>> >>> On Wed, Feb 3, 2016 at 2:11 PM, vfclists . wrote: >>> >>>> What is the syntax for accessing caller profile fields in the dialplan, >>>> for instance if you want to set a channel variable to a caller profile >>>> field? >>>> >>>> Is there also a way of performing some string manipulation and >>>> extraction on the variables in the XML dialplan? >>>> >>>> This is what I am trying to achieve. The CDR from a service provider >>>> contains the CLI from a gateway on the customers premises, but it doesn't >>>> show which extension on the customer's premises the call came from. What I >>>> need is to be able to obtain the extension of the sip device and combine >>>> that with the customer gateway's CLI so that it shows in the CDR record. >>>> >>>> eg if the gateway's CLI is 2340 and the extension of the caller is >>>> 1001, corresponding to the UserId on a sipura, I want the CLI passed to the >>>> service provider to be 23401001. When I check the XML cdr in Freeswitch I >>>> see an XML value which in XPath would be accessed as >>>> /callflow/caller_profile/caller_id_number. All the information is therefore >>>> passed onto Freeswitch for use in the call, and what I need is to be able >>>> to access it and change the effective_caller_id_number before bridging the >>>> call. >>>> >>>> -- >>>> Frank Church >>>> >>>> ======================= >>>> http://devblog.brahmancreations.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> ?talo Rossi >>> italo at freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Frank Church >> >> ======================= >> http://devblog.brahmancreations.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/01a54f18/attachment.html From sos at sokhapkin.dyndns.org Thu Feb 4 00:48:17 2016 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 03 Feb 2016 16:48:17 -0500 Subject: [Freeswitch-users] Is there a way to deal with "false ring-back than 503" carriers? In-Reply-To: <56B25E01.6030608@gmail.com> References: <56B25E01.6030608@gmail.com> Message-ID: <1505680.nGOqxEsikt@sos> You can't continue in bypass media mode because (wrong) media IP/port already sent to your downstream. Even if the second gateway completes the call OK, you will get one way audio. So either proxy the media or fail the call. On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote: > Hello, > > A downstream carrier would accept INVITE, send 183 w/SDP, and almost > instantaneously respond "503". This kills the call not letting re-route > to the next carrier. > > Without calling any specific names, this seems to become more common. > Unfortunately, often the offending carrier is a few steps down the call > path and I can't reach them with a cluebat. > > Is there a way to make FS not fail the call and try the next route if > using LCR as per below? Note that we are in bypass_media mode: > > > > > Or to keep it simple, forget LCR for a moment, can we make it go to > gateway_2 in the following example if gw1 returns "183" followed by "503"? > > > data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destinat > ion_number}"/> > > > Many thanks, > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From victor.chukalovskiy at gmail.com Thu Feb 4 00:58:09 2016 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 3 Feb 2016 16:58:09 -0500 Subject: [Freeswitch-users] Is there a way to deal with "false ring-back than 503" carriers? In-Reply-To: <1505680.nGOqxEsikt@sos> References: <56B25E01.6030608@gmail.com> <1505680.nGOqxEsikt@sos> Message-ID: <56B277F1.5080906@gmail.com> Thank you! Hypothesising there is a way to ontinue to #2 while in bypass_media, as soon as I get 183 SDP from the 2nd carrier, wouldn't it override bogus 183 SDP from the first carrier? It's not an ideal scenario to have two distinct 183's on the same call, but should be doable, no? In proxy_media, is there an additional param or var I'd need in order to continue to #2? Would that be ignore_early_media, or something else? On 16-02-03 04:48 PM, Sergey Okhapkin wrote: > You can't continue in bypass media mode because (wrong) media IP/port already > sent to your downstream. Even if the second gateway completes the call OK, you > will get one way audio. > > So either proxy the media or fail the call. > > On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote: >> Hello, >> >> A downstream carrier would accept INVITE, send 183 w/SDP, and almost >> instantaneously respond "503". This kills the call not letting re-route >> to the next carrier. >> >> Without calling any specific names, this seems to become more common. >> Unfortunately, often the offending carrier is a few steps down the call >> path and I can't reach them with a cluebat. >> >> Is there a way to make FS not fail the call and try the next route if >> using LCR as per below? Note that we are in bypass_media mode: >> >> >> >> >> Or to keep it simple, forget LCR for a moment, can we make it go to >> gateway_2 in the following example if gw1 returns "183" followed by "503"? >> >> >> > data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destinat >> ion_number}"/> >> >> >> Many thanks, >> -Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sos at sokhapkin.dyndns.org Thu Feb 4 01:05:37 2016 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 03 Feb 2016 17:05:37 -0500 Subject: [Freeswitch-users] Is there a way to deal with "false ring-back than 503" carriers? In-Reply-To: <56B277F1.5080906@gmail.com> References: <56B25E01.6030608@gmail.com> <1505680.nGOqxEsikt@sos> <56B277F1.5080906@gmail.com> Message-ID: <3151009.2tZGWi1qvm@sos> Unfortunately 183 from the second gateway will not override the first one. It's how SIP offer/answer works. Once media IP provided to calling party, it can't be changed. Set "continue_on_fail=true" and do not set bypass_media. On Wednesday 03 February 2016 16:58:09 Victor Chukalovskiy wrote: > Thank you! Hypothesising there is a way to ontinue to #2 while in > bypass_media, as soon as I get 183 SDP from the 2nd carrier, wouldn't it > override bogus 183 SDP from the first carrier? It's not an ideal > scenario to have two distinct 183's on the same call, but should be > doable, no? > > In proxy_media, is there an additional param or var I'd need in order to > continue to #2? Would that be ignore_early_media, or something else? > > On 16-02-03 04:48 PM, Sergey Okhapkin wrote: > > You can't continue in bypass media mode because (wrong) media IP/port > > already sent to your downstream. Even if the second gateway completes the > > call OK, you will get one way audio. > > > > So either proxy the media or fail the call. > > > > On Wednesday 03 February 2016 15:07:29 Victor Chukalovskiy wrote: > >> Hello, > >> > >> A downstream carrier would accept INVITE, send 183 w/SDP, and almost > >> instantaneously respond "503". This kills the call not letting re-route > >> to the next carrier. > >> > >> Without calling any specific names, this seems to become more common. > >> Unfortunately, often the offending carrier is a few steps down the call > >> path and I can't reach them with a cluebat. > >> > >> Is there a way to make FS not fail the call and try the next route if > >> using LCR as per below? Note that we are in bypass_media mode: > >> > >> > >> >> data="lcr/lcr_profile/${destination_number}"/> > >> > >> Or to keep it simple, forget LCR for a moment, can we make it go to > >> gateway_2 in the following example if gw1 returns "183" followed by > >> "503"? > >> > >> > >> >> data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${desti > >> nat ion_number}"/> > >> > >> > >> Many thanks, > >> -Victor > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From blackc2004 at gmail.com Thu Feb 4 01:13:22 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 3 Feb 2016 14:13:22 -0800 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN Message-ID: Hi, I've been having a very strange issue lately with loading the core codecs when using core-db-dsn on 1.6.6 and 1.7 on Debian. As long as I am using sqlite, everything appears to work correctly but once I switch to using postgres the core codecs load but don't appear in the show codecs. uname -a Linux c.mydomain.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux Freeswitch start: http://pastebin.com/jD1GLFAF It appears that they load Show codecs after start: http://pastebin.com/W1VfsQDr And the Core codecs are missing Reload the core_pcm_module and show codecs: http://pastebin.com/qNpD0Due And you can see that they are now there. Any help here would be greatly appreciated. Hopefully I'm just missing something small in my setups. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/f586f021/attachment.html From krice at freeswitch.org Thu Feb 4 02:02:32 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 3 Feb 2016 17:02:32 -0600 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: References: Message-ID: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> Are you sure they aren?t actually loading? Show commands only show you whats in the database? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B Sent: Wednesday, February 3, 2016 4:13 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN Hi, I've been having a very strange issue lately with loading the core codecs when using core-db-dsn on 1.6.6 and 1.7 on Debian. As long as I am using sqlite, everything appears to work correctly but once I switch to using postgres the core codecs load but don't appear in the show codecs. uname -a Linux c.mydomain.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux Freeswitch start: http://pastebin.com/jD1GLFAF It appears that they load Show codecs after start: http://pastebin.com/W1VfsQDr And the Core codecs are missing Reload the core_pcm_module and show codecs: http://pastebin.com/qNpD0Due And you can see that they are now there. Any help here would be greatly appreciated. Hopefully I'm just missing something small in my setups. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/beff1516/attachment.html From blackc2004 at gmail.com Thu Feb 4 02:06:49 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 3 Feb 2016 15:06:49 -0800 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> Message-ID: On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice wrote: > Are you sure they aren?t actually loading? Show commands only show you > whats in the database? > > Hi Ken, How can I confirm that? When I try to place a call before running the reload command I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/247891be/attachment.html From krice at freeswitch.org Thu Feb 4 02:30:52 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 3 Feb 2016 17:30:52 -0600 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> Message-ID: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> The show commands only do a select on the database and print that out to screen? all the show commands work this way? Sounds like you have something delaying the start of the coredb? are you doing something like trying to load configs for the core from xml_curl? As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B Sent: Wednesday, February 3, 2016 5:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice > wrote: Are you sure they aren?t actually loading? Show commands only show you whats in the database? Hi Ken, How can I confirm that? When I try to place a call before running the reload command I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/a368b536/attachment-0001.html From max at nysolutions.com Thu Feb 4 04:46:27 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 4 Feb 2016 01:46:27 +0000 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> Message-ID: <1964d5f4fef0424583e184a4b030635b@nysolutions.com> I saw someone in the Fusion IRC channel with the same issue they said it was OK after doing a reload CORE_PCM_MODULE Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, February 3, 2016 6:31 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN The show commands only do a select on the database and print that out to screen? all the show commands work this way? Sounds like you have something delaying the start of the coredb? are you doing something like trying to load configs for the core from xml_curl? As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B Sent: Wednesday, February 3, 2016 5:07 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice > wrote: Are you sure they aren?t actually loading? Show commands only show you whats in the database? Hi Ken, How can I confirm that? When I try to place a call before running the reload command I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/f8bdfbf8/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/f8bdfbf8/attachment.jpg From blackc2004 at gmail.com Thu Feb 4 05:48:06 2016 From: blackc2004 at gmail.com (Cj B) Date: Wed, 3 Feb 2016 18:48:06 -0800 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> Message-ID: <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com> I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA You are right though that even though show codecs isn?t displaying the codecs, it appears to at least be matching them from the phone. Now that you?ve pointed that out, it?s looking more like it?s not converting the call to PCMU/PCMA for the provider? Thanks > On Feb 3, 2016, at 3:30 PM, Ken Rice wrote: > > The show commands only do a select on the database and print that out to screen? all the show commands work this way? > > Sounds like you have something delaying the start of the coredb? are you doing something like trying to load configs for the core from xml_curl? > > As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cj B > Sent: Wednesday, February 3, 2016 5:07 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN > > > On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice > wrote: >> Are you sure they aren?t actually loading? Show commands only show you whats in the database? > > Hi Ken, How can I confirm that? When I try to place a call before running the reload command I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160203/1daccd10/attachment-0001.html From vfclists at gmail.com Thu Feb 4 13:37:50 2016 From: vfclists at gmail.com (vfclists .) Date: Thu, 4 Feb 2016 10:37:50 +0000 Subject: [Freeswitch-users] How do you check the settings a Freeswitch session was started with? Message-ID: My Freeswitch is configured to start at runtime, but doesn't seem to work until I stop it, switch to the /usr/local/freeswitch/conf directory, and start it using '/usr/local/freeswitch/bin/freeswitch -nc'. When I copy those commands to '/etc/rc.local', it still doesn't work ie cd /usr/local/freeswitch/conf /usr/local/freeswitch/bin/freeswitch -nc I think the service script running it is not configured correctly. Is there some fs_cli, or fsctl option that can be used to interrogate a running instance to see what settings it was started with, or what configuration files it is using? -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/0c781c6b/attachment.html From bote_radio at botecomm.com Thu Feb 4 17:59:12 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 4 Feb 2016 09:59:12 -0500 Subject: [Freeswitch-users] How do you check the settings a Freeswitch session was started with? In-Reply-To: References: Message-ID: <00db01d15f5c$9bef5b80$d3ce1280$@botecomm.com> The world is moving to ?systemd? to initialize daemons and applications. There are some notes about systemd on this deprecated Confluence page, but we really should figure out a dedicated documentation of systemd. Probably from the Stash source tree, I?m thinking, just to put it out there. https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video If you?re using sysVinit then maybe Google will turn up an older version of an init script. I think you should be looking in the o.s. logs to see what it thinks is happening when it starts FS. I think Debian stores this stuff in /var/log/access or /var/log/process. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: vfclists . Sent: Thursday, 04 February, 2016 05:38 Subject: [Freeswitch-users] How do you check the settings a Freeswitch session was started with? My Freeswitch is configured to start at runtime, but doesn't seem to work until I stop it, switch to the /usr/local/freeswitch/conf directory, and start it using '/usr/local/freeswitch/bin/freeswitch -nc'. When I copy those commands to '/etc/rc.local', it still doesn't work ie cd /usr/local/freeswitch/conf /usr/local/freeswitch/bin/freeswitch -nc I think the service script running it is not configured correctly. Is there some fs_cli, or fsctl option that can be used to interrogate a running instance to see what settings it was started with, or what configuration files it is using? -- Frank Church ======================= http://devblog.brahmancreations.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/31e03e81/attachment.html From decipher.hk at gmail.com Thu Feb 4 03:35:12 2016 From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=) Date: Thu, 04 Feb 2016 00:35:12 +0000 Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter Message-ID: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com> Hello everyone!, I'm testing a mod_callcenter for make a FreeSWITCH version of a open source software (https://github.com/roramirez/qpanel/tree/fs) Now i using ESL to send command and get information from the module. I looking for a way to know the data of answered/abandoned call from a queue and agents. Somebody can give a tips or a light? Regards, -- Rodrigo Ram?rez Norambuena http://www.rodrigoramirez.com From tim.compnetwork at gmail.com Thu Feb 4 19:19:18 2016 From: tim.compnetwork at gmail.com (Tim King) Date: Thu, 4 Feb 2016 11:19:18 -0500 Subject: [Freeswitch-users] X-Auth-IP Variable? Message-ID: I am using proxy authentication in my setup and it is working. To do this I have created an acl *autoload_configs/acl.conf.xml* *sip_profiles/external.xml* This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan. This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/bc43892b/attachment.html From roman at dissauer.net Thu Feb 4 19:27:18 2016 From: roman at dissauer.net (Roman Dissauer) Date: Thu, 4 Feb 2016 17:27:18 +0100 Subject: [Freeswitch-users] BLF for Gateway In-Reply-To: References: Message-ID: <7A4B9B3A-ABCB-493C-A9D3-CB871EE6CC73@dissauer.net> Thanks, I?ll investigate on that and post my findings. Best Regards, Roman > Am 03.02.2016 um 17:45 schrieb Michael Jerris : > > You might be able to do something here with display updates. > >> On Feb 2, 2016, at 6:58 AM, Roman Dissauer wrote: >> >> Hi All, >> >> is there a way to get gateway usage in freeswitch on my phones blf? >> I have multiple gateways registered and want to see which one is taken for a particular outbound call. >> >> Best Regards, >> Roman > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at cassidywebservices.co.uk Thu Feb 4 19:44:43 2016 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 4 Feb 2016 16:44:43 +0000 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: Message-ID: ${variable_sip_h_X-Auth-IP} On 4 February 2016 at 16:19, Tim King wrote: > I am using proxy authentication in my setup and it is working. To do this > I have created an acl > *autoload_configs/acl.conf.xml* > > > > > > > *sip_profiles/external.xml* > > > > > > This is all working as desired. The problem is prior to adding the > opensips I was using the network_addr variable in my dialplan. > > > expression="true" break="on-false"/> > expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> > data="dtmf_type=rfc2833"/> > data="accountcode=customer1123"/> > data="continue_on_fail=false"/> > data="hangup_after_bridge=true"/> > data="proxy_media=true"/> > data="sofia/external/$2 at 8.7.6.5:5060"/> > > > > This of course no longer works because network_addr is always the address > of my proxy server. How can I get the address from the X-authip into the > dialplan? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/83898a7e/attachment-0001.html From tim.compnetwork at gmail.com Thu Feb 4 20:13:24 2016 From: tim.compnetwork at gmail.com (Tim King) Date: Thu, 4 Feb 2016 12:13:24 -0500 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: Message-ID: Thank you for the reply. I tried this for matching to the ACL but it is failing. On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > ${variable_sip_h_X-Auth-IP} > > On 4 February 2016 at 16:19, Tim King wrote: > >> I am using proxy authentication in my setup and it is working. To do this >> I have created an acl >> *autoload_configs/acl.conf.xml* >> >> >> >> >> >> >> *sip_profiles/external.xml* >> >> >> >> >> >> This is all working as desired. The problem is prior to adding the >> opensips I was using the network_addr variable in my dialplan. >> >> >> > expression="true" break="on-false"/> >> > expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> >> > data="dtmf_type=rfc2833"/> >> > data="accountcode=customer1123"/> >> > data="continue_on_fail=false"/> >> > data="hangup_after_bridge=true"/> >> > data="proxy_media=true"/> >> > data="sofia/external/$2 at 8.7.6.5:5060"/> >> >> >> >> This of course no longer works because network_addr is always the address >> of my proxy server. How can I get the address from the X-authip into the >> dialplan? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/5e95e681/attachment.html From andrew at cassidywebservices.co.uk Thu Feb 4 20:23:01 2016 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 4 Feb 2016 17:23:01 +0000 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: Message-ID: Can you send a sip trace, particularly the one that should have the X-Auth-IP set? Thanks, On 4 February 2016 at 17:13, Tim King wrote: > Thank you for the reply. I tried this for matching to the ACL but it is > failing. > expression="true" break="on-false"/> > > On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> ${variable_sip_h_X-Auth-IP} >> >> On 4 February 2016 at 16:19, Tim King wrote: >> >>> I am using proxy authentication in my setup and it is working. To do >>> this I have created an acl >>> *autoload_configs/acl.conf.xml* >>> >>> >>> >>> >>> >>> >>> *sip_profiles/external.xml* >>> >>> >>> >>> >>> >>> This is all working as desired. The problem is prior to adding the >>> opensips I was using the network_addr variable in my dialplan. >>> >>> >>> >> expression="true" break="on-false"/> >>> >> expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> >>> >> data="dtmf_type=rfc2833"/> >>> >> data="accountcode=customer1123"/> >>> >> data="continue_on_fail=false"/> >>> >> data="hangup_after_bridge=true"/> >>> >> data="proxy_media=true"/> >>> >> data="sofia/external/$2 at 8.7.6.5:5060"/> >>> >>> >>> >>> This of course no longer works because network_addr is always the >>> address of my proxy server. How can I get the address from the X-authip >>> into the dialplan? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/92da9f4c/attachment-0001.html From bilaln018 at gmail.com Thu Feb 4 20:25:00 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 4 Feb 2016 22:25:00 +0500 Subject: [Freeswitch-users] [Session timer too low][SIP 422] Message-ID: Hi users, I am integrating FreeSwitch with web-sockets, using http://tryit.jssip.net/. So my problem is the extension got registered, but after that when i dial a number i am getting "SIP FAILURE CODE" on JsSIP, i took a network trace, and its showing me SIP 422 responce to invite "Sesion interval too low", now i understand that i need to increase the value of Session Expires, but i cant increase that as it what is configured in tryjssip, so can i some how disable this check from freeswicth? As i guess 90 is the border value that RFC gave. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/2a728039/attachment.html From mic.niel84 at gmail.com Thu Feb 4 20:28:05 2016 From: mic.niel84 at gmail.com (Michael Nielsen) Date: Thu, 4 Feb 2016 18:28:05 +0100 Subject: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel In-Reply-To: <004401d15b96$7274e320$575ea960$@botecomm.com> References: <009c01d15333$b95e5660$2c1b0320$@botecomm.com> <004401d15b96$7274e320$575ea960$@botecomm.com> Message-ID: Would that be a LUA script? On Saturday, January 30, 2016, Bote Man wrote: > Perhaps your routing is getting complex enough to justify writing a script > to do the parsing and routing? I don?t know, just suggesting that as the > XML dialplan does have its limitations. It?s entirely possible that the XML > dialplan can do it and I simply don?t know what it is; I am by no means a > wizard at this. > > > > --- > > Bote > > > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > *From:* Michael Nielsen > *Sent:* Saturday, 30 January, 2016 03:23 > *Subject:* Re: [Freeswitch-users] loopback/XYZ doesn't contain variables > from channel > > > > The reason for using loopback is that I'm not sure if the dialed number is > and external or internal number. > > The auto dialed number is dynamic and can be everyone from 1 number to > many numbers. > > > > Both internal and external numbers all matches real phone numbers ex. > +44223849591. > > > > So I have another dial plan which examinate numbers and route them > accordingly - and appends country code if necessary etc. > > > On Wednesday, January 20, 2016, Bote Man > wrote: > > I?m not sure that you need to use the loopback special channel to > accomplish this. What happens if you simply set > > > > > > > > > > ?and let FreeSWITCH do its normal routing? I think the loopback might be > complicating matters. The above lines work for me, anyway. > > > > I think under the hood the auto_outcall simply stacks up originate > commands so if the plain outside telephone number doesn?t work as-is, try > sofia/gateway-name-here/3438773 or variations on that syntax to see if > that works. > > > > There?s a possibility that you?re overthinking this with the loopback > channel* J > > > > Hope this helps. > > > > * We must always keep in mind that loopback is evil. Amen. > > > > --- > > Bote > > > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > *From:* Michael Nielsen > *Sent:* Tuesday, 19 January, 2016 10:51 > *Subject:* Re: [Freeswitch-users] loopback/XYZ doesn't contain variables > from channel > > > > My reason for this question is that I'm using the conference module, and > are trying to data="loopback/1001,loopback/1002,loopback/3438773"/> > > > > 2 of the called numbers should be handled internally and 1 should be going > through my sip provider. > > So I need all 3 of them to be handled from the top of my dial plan - with > the correct variables set. > > > > > > > > On Fri, Jan 15, 2016 at 2:16 PM, Michael Nielsen > wrote: > > I've tried that as well. > > CLI shows: > > Dialplan: sofia/internal/+44234987447 at my-domain Action > set(loopback_export=country_code) > > After the loopback the dialplan is as follows: > > Dialplan: loopback/32487477-b Action > set(dialed_number=${country_code} 32487477) INLINE > > EXECUTE loopback/32487477-b set(dialed_number=32487477) > > So +44 is not added, even though the country_code variable is +44 before > the loopback. > > > > > > On Fri, Jan 15, 2016 at 2:10 PM, Michael Jerris wrote: > > > > > > On Friday, January 15, 2016, Michael Nielsen wrote: > > I've tried to add: > > data="loopback_export=[country_code=${country_code}]"/> > > > > And my log shows the following: > > set(loopback_export=[country_code=+44]) > > But it still isn't available on the next runthrough of dial plan. > > > > On Wed, Jan 13, 2016 at 6:51 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > set > > loopback_export to the names of all of the variables you want to copy > across sep buy a , (comma) > > > > > > On Wed, Jan 13, 2016 at 8:23 AM, Michael Nielsen > wrote: > > I've got directory with subscribers containing a variable called > "country_code". > > This variable can be used in regular dial plans and works perfectly. > > > > However, I've got a conference call dial plan where I > use conference_set_auto_outcall. > > I'mm calling loopback/XYZ to get my other dial plans into play. > > > > Everything works, except my subscriber variable "country_code" doesn't get > recognised after the loopback/. > > > > How can I fix this kind of issue? > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/e6113b70/attachment.html From bote_radio at botecomm.com Thu Feb 4 20:45:17 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 4 Feb 2016 12:45:17 -0500 Subject: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel In-Reply-To: References: <009c01d15333$b95e5660$2c1b0320$@botecomm.com> <004401d15b96$7274e320$575ea960$@botecomm.com> Message-ID: <003701d15f73$cfeae2d0$6fc0a870$@botecomm.com> Lua, perl, whatever is your strongest suit. You should not do telephony in the script, just figure out the best route to the destination and present that to FreeSWITCH to do the heavy lifting of telephony signaling and media setup. This is one solution that springs to mind, but perhaps the XML dialplan can do it, I don?t know. Bote From: Michael Nielsen Sent: Thursday, 04 February, 2016 12:28 Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel Would that be a LUA script? On Saturday, January 30, 2016, Bote Man wrote: Perhaps your routing is getting complex enough to justify writing a script to do the parsing and routing? I don?t know, just suggesting that as the XML dialplan does have its limitations. It?s entirely possible that the XML dialplan can do it and I simply don?t know what it is; I am by no means a wizard at this. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: Michael Nielsen Sent: Saturday, 30 January, 2016 03:23 Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel The reason for using loopback is that I'm not sure if the dialed number is and external or internal number. The auto dialed number is dynamic and can be everyone from 1 number to many numbers. Both internal and external numbers all matches real phone numbers ex. +44223849591. So I have another dial plan which examinate numbers and route them accordingly - and appends country code if necessary etc. On Wednesday, January 20, 2016, Bote Man > wrote: I?m not sure that you need to use the loopback special channel to accomplish this. What happens if you simply set ?and let FreeSWITCH do its normal routing? I think the loopback might be complicating matters. The above lines work for me, anyway. I think under the hood the auto_outcall simply stacks up originate commands so if the plain outside telephone number doesn?t work as-is, try sofia/gateway-name-here/3438773 or variations on that syntax to see if that works. There?s a possibility that you?re overthinking this with the loopback channel* J Hope this helps. * We must always keep in mind that loopback is evil. Amen. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: Michael Nielsen Sent: Tuesday, 19 January, 2016 10:51 Subject: Re: [Freeswitch-users] loopback/XYZ doesn't contain variables from channel My reason for this question is that I'm using the conference module, and are trying to 2 of the called numbers should be handled internally and 1 should be going through my sip provider. So I need all 3 of them to be handled from the top of my dial plan - with the correct variables set. On Fri, Jan 15, 2016 at 2:16 PM, Michael Nielsen wrote: I've tried that as well. CLI shows: Dialplan: sofia/internal/+44234987447 at my-domain Action set(loopback_export=country_code) After the loopback the dialplan is as follows: Dialplan: loopback/32487477-b Action set(dialed_number=${country_code} 32487477) INLINE EXECUTE loopback/32487477-b set(dialed_number=32487477) So +44 is not added, even though the country_code variable is +44 before the loopback. On Fri, Jan 15, 2016 at 2:10 PM, Michael Jerris wrote: On Friday, January 15, 2016, Michael Nielsen wrote: I've tried to add: And my log shows the following: set(loopback_export=[country_code=+44]) But it still isn't available on the next runthrough of dial plan. On Wed, Jan 13, 2016 at 6:51 PM, Anthony Minessale wrote: set loopback_export to the names of all of the variables you want to copy across sep buy a , (comma) On Wed, Jan 13, 2016 at 8:23 AM, Michael Nielsen wrote: I've got directory with subscribers containing a variable called "country_code". This variable can be used in regular dial plans and works perfectly. However, I've got a conference call dial plan where I use conference_set_auto_outcall. I'mm calling loopback/XYZ to get my other dial plans into play. Everything works, except my subscriber variable "country_code" doesn't get recognised after the loopback/. How can I fix this kind of issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/ab4f68c6/attachment-0001.html From bote_radio at botecomm.com Thu Feb 4 20:45:17 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 4 Feb 2016 12:45:17 -0500 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: Message-ID: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com> I also recommend using the FreeSWITCH ?log? application to display what FS thinks those variables contain. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: Tim King Sent: Thursday, 04 February, 2016 12:13 Subject: Re: [Freeswitch-users] X-Auth-IP Variable? Thank you for the reply. I tried this for matching to the ACL but it is failing. On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy wrote: ${variable_sip_h_X-Auth-IP} On 4 February 2016 at 16:19, Tim King wrote: I am using proxy authentication in my setup and it is working. To do this I have created an acl autoload_configs/acl.conf.xml sip_profiles/external.xml This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan. This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan? -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/3b80a9ce/attachment.html From tim.compnetwork at gmail.com Thu Feb 4 20:52:19 2016 From: tim.compnetwork at gmail.com (Tim King) Date: Thu, 4 Feb 2016 12:52:19 -0500 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com> References: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com> Message-ID: I keep trying to log them but the log application is showing me the log statement and not the variable itself. I know I have used this successfully before, but even if I try logging the destination_number it does not log. I have tried ${destination_number} and [${destination_number}] and my log output looks like this. Action log(Log TEST ========== ${destination_number} or [${destination_number}] On Thu, Feb 4, 2016 at 12:45 PM, Bote Man wrote: > I also recommend using the FreeSWITCH ?log? application to display what FS > thinks those variables contain. > > > > > > --- > > Bote > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > *From:* Tim King > *Sent:* Thursday, 04 February, 2016 12:13 > *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable? > > > > Thank you for the reply. I tried this for matching to the ACL but it is > failing. > > expression="true" break="on-false"/> > > > > On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > > ${variable_sip_h_X-Auth-IP} > > > > On 4 February 2016 at 16:19, Tim King wrote: > > I am using proxy authentication in my setup and it is working. To do this > I have created an acl > > *autoload_configs/acl.conf.xml* > > > > > > > > > > > > > > *sip_profiles/external.xml* > > > > > > > > > > > > This is all working as desired. The problem is prior to adding the > opensips I was using the network_addr variable in my dialplan. > > > > > > expression="true" break="on-false"/> > > expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> > > data="dtmf_type=rfc2833"/> > > data="accountcode=customer1123"/> > > data="continue_on_fail=false"/> > > data="hangup_after_bridge=true"/> > > data="proxy_media=true"/> > > data="sofia/external/$2 at 8.7.6.5:5060"/> > > > > > > > > This of course no longer works because network_addr is always the address > of my proxy server. How can I get the address from the X-authip into the > dialplan? > > > > > > > > -- > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > Managing Director > > > > *T *03300 100 960 *F > *03300 100 961 > > *E *andrew at cassidywebservices.co.uk > > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/fe53d9d4/attachment-0001.html From blackc2004 at gmail.com Thu Feb 4 21:02:33 2016 From: blackc2004 at gmail.com (Cj B) Date: Thu, 4 Feb 2016 10:02:33 -0800 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com> References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com> Message-ID: Good Morning, I worked with the fusion guys and found that if I set absolut_codec_string=pcma on the outbound routes the calls are completing ok. I also have inbound-codec-negotiation=generous on my sip profiles, maybe I need to change that to scrooge or something else? Any other suggestions? Thanks > On Feb 3, 2016, at 6:48 PM, Cj B wrote: > > I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA > > You are right though that even though show codecs isn?t displaying the codecs, it appears to at least be matching them from the phone. Now that you?ve pointed that out, it?s looking more like it?s not converting the call to PCMU/PCMA for the provider? > > Thanks > >> On Feb 3, 2016, at 3:30 PM, Ken Rice > wrote: >> >> The show commands only do a select on the database and print that out to screen? all the show commands work this way? >> >> Sounds like you have something delaying the start of the coredb? are you doing something like trying to load configs for the core from xml_curl? >> >> As far as why are you getting incompatible destination during media negotiation, post a FULL unedited call log with sip trace enabled of this scenario to the paste bin or ask on IRC someone to review your call log? the underlying error should be readily appearent there >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Cj B >> Sent: Wednesday, February 3, 2016 5:07 PM >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN >> >> >> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice > wrote: >>> Are you sure they aren?t actually loading? Show commands only show you whats in the database? >> >> Hi Ken, How can I confirm that? When I try to place a call before running the reload command I get an error about [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it seems like they are either not loading properly or they are loading before the database? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/9ddaeb96/attachment.html From brian at freeswitch.org Thu Feb 4 21:05:39 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Feb 2016 12:05:39 -0600 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com> Message-ID: The codec is loading and your do config is broken during start up so it may not appear in show modules in this case, we may need to have a JIRA filed in this so we can investigate and get the facts On Thursday, February 4, 2016, Cj B wrote: > Good Morning, > > I worked with the fusion guys and found that if I set > absolut_codec_string=pcma on the outbound routes the calls are completing > ok. I also have inbound-codec-negotiation=generous on my sip profiles, > maybe I need to change that to scrooge or something else? > > Any other suggestions? > > Thanks > > On Feb 3, 2016, at 6:48 PM, Cj B > wrote: > > I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. > Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA > > You are right though that even though show codecs isn?t displaying the > codecs, it appears to at least be matching them from the phone. Now that > you?ve pointed that out, it?s looking more like it?s not converting the > call to PCMU/PCMA for the provider? > > Thanks > > On Feb 3, 2016, at 3:30 PM, Ken Rice > wrote: > > The show commands only do a select on the database and print that out to > screen? all the show commands work this way? > > Sounds like you have something delaying the start of the coredb? are you > doing something like trying to load configs for the core from xml_curl? > > As far as why are you getting incompatible destination during media > negotiation, post a FULL unedited call log with sip trace enabled of this > scenario to the paste bin or ask on IRC someone to review your call log? > the underlying error should be readily appearent there > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > > ] *On Behalf Of *Cj B > *Sent:* Wednesday, February 3, 2016 5:07 PM > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load > correctly with DSN > > > On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice > wrote: > > Are you sure they aren?t actually loading? Show commands only show you > whats in the database? > > > Hi Ken, How can I confirm that? When I try to place a call before running > the reload command I get an error about [CS_CONSUME_MEDIA] > [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it > seems like they are either not loading properly or they are loading before > the database? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/56cf04cf/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 4 23:08:10 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Feb 2016 14:08:10 -0600 Subject: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load correctly with DSN In-Reply-To: References: <5ca701d15ed6$f7596be0$e60c43a0$@freeswitch.org> <5cbe01d15eda$ecb39a90$c61acfb0$@freeswitch.org> <68F25448-68E7-4FD7-A6BE-613DDE2D4A2A@gmail.com> Message-ID: Look for errors in the SQL server logs right after a fresh restart. On Thu, Feb 4, 2016 at 12:05 PM, Brian West wrote: > The codec is loading and your do config is broken during start up so it > may not appear in show modules in this case, we may need to have a JIRA > filed in this so we can investigate and get the facts > > > On Thursday, February 4, 2016, Cj B wrote: > >> Good Morning, >> >> I worked with the fusion guys and found that if I set >> absolut_codec_string=pcma on the outbound routes the calls are completing >> ok. I also have inbound-codec-negotiation=generous on my sip profiles, >> maybe I need to change that to scrooge or something else? >> >> Any other suggestions? >> >> Thanks >> >> On Feb 3, 2016, at 6:48 PM, Cj B wrote: >> >> I?m using FusionPBX so I?m not sure exactly how it?s loading the configs. >> Here?s an unedited call log of the call: http://pastebin.com/nu3964ZA >> >> You are right though that even though show codecs isn?t displaying the >> codecs, it appears to at least be matching them from the phone. Now that >> you?ve pointed that out, it?s looking more like it?s not converting the >> call to PCMU/PCMA for the provider? >> >> Thanks >> >> On Feb 3, 2016, at 3:30 PM, Ken Rice wrote: >> >> The show commands only do a select on the database and print that out to >> screen? all the show commands work this way? >> >> Sounds like you have something delaying the start of the coredb? are you >> doing something like trying to load configs for the core from xml_curl? >> >> As far as why are you getting incompatible destination during media >> negotiation, post a FULL unedited call log with sip trace enabled of this >> scenario to the paste bin or ask on IRC someone to review your call log? >> the underlying error should be readily appearent there >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [ >> mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cj B >> *Sent:* Wednesday, February 3, 2016 5:07 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] CORE_PCM_MODULE doesn't seem to load >> correctly with DSN >> >> >> On Wed, Feb 3, 2016 at 3:02 PM, Ken Rice wrote: >> >> Are you sure they aren?t actually loading? Show commands only show you >> whats in the database? >> >> >> Hi Ken, How can I confirm that? When I try to place a call before running >> the reload command I get an error about [CS_CONSUME_MEDIA] >> [INCOMPATIBLE_DESTINATION] after reloading the the calls complete, so it >> seems like they are either not loading properly or they are loading before >> the database? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/e9916ba0/attachment.html From lists at telefaks.de Fri Feb 5 00:20:26 2016 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 04 Feb 2016 22:20:26 +0100 Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt Message-ID: <56B3C09A.60907@telefaks.de> I have inherited a Freeswitch installation on a Windows server, which I remotely manage with cygwin and ssh. This worked more or less nicely so far. But today I cannot manage Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no change. I can run fs_cli.exe, and I see the Freeswitch messages, but I do __not__ get a prompt. When I try to enter characters, nothing happens. On the machine itself, when I open a console and start cygwin, fs_cli.exe works as expected with prompt. But not via ssh. But fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be fine in general. What I understand is, that fs_cli connects to Freeswitch via port 8021. So when seeing FS messages, I expect that the connection is already fine, right? I also tried to connect via telnet to port 8021. This works in the local cygwin shell (get an auth request), but not via remote ssh shell (telnet terminates without message). I am puzzled. Anybody has a clue, why I do not have a prompt and where to search? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From peterg at sytelco.com Thu Feb 4 22:42:36 2016 From: peterg at sytelco.com (Piotr Gregor) Date: Thu, 4 Feb 2016 19:42:36 +0000 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? In-Reply-To: References: Message-ID: Hi, "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. Am I correct about that? " Yes. The AVMD module works by calculating the estimate of the frequency and amplitude of signal using DESA-2 algorithm. It fires amd::beep event when the variance for the frequency estimate is below the threshold. The event is of form: Event-Subclass: avmd%3A%3Abeep Event-Name: CUSTOM Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 FreeSWITCH-Hostname: home FreeSWITCH-Switchname: home FreeSWITCH-IPv4: 128.11.35.8 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2016-02-04%2016%3A56%3A55 Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT Event-Date-Timestamp: 1454605015915799 Event-Calling-File: mod_avmd.c Event-Calling-Function: avmd_process Event-Calling-Line-Number: 556 Event-Sequence: 950 Beep-Status: stop Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a call-command: avmd The AVMD doesn't take into consideration segments of speech and segments of silence. It simply calculates it's estimate of frequency and amplitude. "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? " It's use is to detect beep only. You can test how it works by creating extension that will play a tone of given frequency, e.g: This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms. Start avmd on the call with fs_cli: avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start and inspect events by regestering to avmd events with: events plain CUSTOM avmd::beep Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments. You can do this by subscription to TALK/NOTALK events. Hope this helps. cheers, Piotr On 4 February 2016 at 17:50, Piotr Gregor wrote: > Hi, > "I looked at the AVMD module documentation at > https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it > seems to indicate that this simply detects a beep, i.e. it does not use > talking / silence heuristics into account to determine if the call is > answered by a human or machine. > Am I correct about that? " > > Yes. The AVMD module works by calculating the estimate of the frequency > using DESA-2 algorithm. It fires amd::beep event when the variance for that > estimate is below threshold. > The event is of form: > > Event-Subclass: avmd%3A%3Abeep > Event-Name: CUSTOM > Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 > FreeSWITCH-Hostname: home > FreeSWITCH-Switchname: home > FreeSWITCH-IPv4: 128.11.35.8 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2016-02-04%2016%3A56%3A55 > Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT > Event-Date-Timestamp: 1454605015915799 > Event-Calling-File: mod_avmd.c > Event-Calling-Function: avmd_process > Event-Calling-Line-Number: 556 > Event-Sequence: 950 > Beep-Status: stop > Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a > call-command: avmd > > The AVMD doesn't take into consideration TALK/NOTALK events. > > > "How is this intended to be used, do I assume that the call is answered > by a human until this module sends a avmd%3A%3Abeep event to me? How do > others use this module or other techniques to determine human / machine > answer to outbound calls with Freeswitch? " > > You can test how it works by creating extension that will play a tone of > given frequency, e.g: > > > data="tone_stream://L=3;%(500,6850,850)" /> > > > > This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 > ms. > > Start avmd on the call with fs_cli: > avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start > > and inspect events by regestering to avmd events with: > events plain CUSTOM avmd::beep > > Analysing the audio for a presence of the tone is not enough for answering > machine detection. You should also analyse at least the length of > speech/silence segments. > Hope this helps. > > cheers, > Piotr > > > > > > > > On 3 February 2016 at 19:50, Christopher Rienzo > wrote: > >> I wouldn't use beep detection in a dialer application, but it could be >> useful in something like follow me to reduce the occurrence of voicemails >> being left on subscriber phones. >> >> On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig > > wrote: >> >>> I have a client that needs to reliably detect if their outbound calls >>> are answered by a human or voicemail system, so that they can take >>> different actions based on that determination. >>> >>> I looked at the AVMD module documentation at >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it >>> seems to indicate that this simply detects a beep, i.e. it does not use >>> talking / silence heuristics into account to determine if the call is >>> answered by a human or machine. >>> >>> Am I correct about that? How is this intended to be used, do I assume >>> that the call is answered by a human until this module sends >>> a avmd%3A%3Abeep event to me? How do others use this module or other >>> techniques to determine human / machine answer to outbound calls with >>> Freeswitch? >>> >>> Thanks! >>> Bob >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/adbd1967/attachment-0001.html From dre at imerchantsystems.com Thu Feb 4 23:29:33 2016 From: dre at imerchantsystems.com (Londre) Date: Thu, 4 Feb 2016 15:29:33 -0500 Subject: [Freeswitch-users] mod_xml_curl Message-ID: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com> Hi, I have read the wiki several times and followed all the steps to the best of my knowledge. I have gotten mod_xml_curl build and making requests. I set xml_curl debug_on and I can see the request that freeswitch sends and I can take a look at the file it generates in the /tmp/ directory. However the file doesn't seem to be writing anything. I have my web service set up to hand a POST request but it looks like freeswitch is sending GET requests even though I set the method to POST in xml_curl.xml. Any ideal would could be going wrong? System Info: Debian Jessie Freeswitch 1.7 built from source. Xml_curl.xml: 1 2 3 4 5 6 7 8 9 It also doesn't seem to be sending the variables for the gateway-credentials. Its seems weird that its sending a GET request when the default should be a POST. Also it is a little confusing on the wiki some things show a query string and other show header variables. I'd prefer to use POST. Please offer some insight into what could be going wrong. Londre Blocker Developer iMerchant Systems Inc. T: 844.727.1998 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/18db7d81/attachment-0001.html From bote_radio at botecomm.com Fri Feb 5 00:44:23 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 4 Feb 2016 16:44:23 -0500 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com> Message-ID: <007701d15f95$365ceb00$a316c100$@botecomm.com> Here?s what works for me: This will display it in fs_cli at least. I?m not familiar with the syntax of what you posted, unless that?s an API command being pumped into FS via ESL or something. Bote From: Tim King Sent: Thursday, 04 February, 2016 12:52 Subject: Re: [Freeswitch-users] X-Auth-IP Variable? I keep trying to log them but the log application is showing me the log statement and not the variable itself. I know I have used this successfully before, but even if I try logging the destination_number it does not log. I have tried ${destination_number} and [${destination_number}] and my log output looks like this. Action log(Log TEST ========== ${destination_number} or [${destination_number}] On Thu, Feb 4, 2016 at 12:45 PM, Bote Man wrote: I also recommend using the FreeSWITCH ?log? application to display what FS thinks those variables contain. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence From: Tim King Sent: Thursday, 04 February, 2016 12:13 Subject: Re: [Freeswitch-users] X-Auth-IP Variable? Thank you for the reply. I tried this for matching to the ACL but it is failing. On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy wrote: ${variable_sip_h_X-Auth-IP} On 4 February 2016 at 16:19, Tim King wrote: I am using proxy authentication in my setup and it is working. To do this I have created an acl autoload_configs/acl.conf.xml sip_profiles/external.xml This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan. This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan? -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/1f6e1705/attachment.html From bote_radio at botecomm.com Fri Feb 5 00:47:29 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 4 Feb 2016 16:47:29 -0500 Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt In-Reply-To: <56B3C09A.60907@telefaks.de> References: <56B3C09A.60907@telefaks.de> Message-ID: <007c01d15f95$a588b220$f09a1660$@botecomm.com> You can only connect to port 8021 locally because Is set in autoload_configs/event_socket.conf.xml If you set that to value to 0.0.0.0 the whole world can connect to port 8021, but the problem with that is that the whole world can connect. I wouldn't change without careful consideration of the security impact. I have never used ssh to connect to Windows machine, I didn't even know that it is possible so that is a new one to me. I just remote desktop (mstsc.exe) into the target machine and run the necessary commands and utilities directly on the box. Sorry I couldn't be more help. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence > -----Original Message----- > From: Peter Steinbach > Sent: Thursday, 04 February, 2016 16:20 > Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt > > I have inherited a Freeswitch installation on a Windows server, which I > remotely manage with cygwin and ssh. > > This worked more or less nicely so far. But today I cannot manage > Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no > change. > I can run fs_cli.exe, and I see the Freeswitch messages, but I do > __not__ get a prompt. When I try to enter characters, nothing happens. > > On the machine itself, when I open a console and start cygwin, > fs_cli.exe works as expected with prompt. But not via ssh. But > fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be > fine in general. > > What I understand is, that fs_cli connects to Freeswitch via port 8021. > So when seeing FS messages, I expect that the connection is already > fine, right? > > I also tried to connect via telnet to port 8021. This works in the local > cygwin shell (get an auth request), but not via remote ssh shell (telnet > terminates without message). > > I am puzzled. Anybody has a clue, why I do not have a prompt and where > to search? > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Feb 5 01:06:25 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Feb 2016 17:06:25 -0500 Subject: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system? In-Reply-To: References: Message-ID: See the previous response about our commercial module available if you are looking for something that detects segments of speech/silence. That is exactly what that module does. > On Feb 4, 2016, at 2:42 PM, Piotr Gregor wrote: > > Hi, > "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. > Am I correct about that? " > > Yes. The AVMD module works by calculating the estimate of the frequency and amplitude of signal using DESA-2 algorithm. It fires amd::beep event when the variance for the frequency estimate is below the threshold. > The event is of form: > > Event-Subclass: avmd%3A%3Abeep > Event-Name: CUSTOM > Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 > FreeSWITCH-Hostname: home > FreeSWITCH-Switchname: home > FreeSWITCH-IPv4: 128.11.35.8 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2016-02-04%2016%3A56%3A55 > Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT > Event-Date-Timestamp: 1454605015915799 > Event-Calling-File: mod_avmd.c > Event-Calling-Function: avmd_process > Event-Calling-Line-Number: 556 > Event-Sequence: 950 > Beep-Status: stop > Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a > call-command: avmd > > The AVMD doesn't take into consideration segments of speech and segments of silence. It simply calculates it's estimate of frequency and amplitude. > > > "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? " > > It's use is to detect beep only. > You can test how it works by creating extension that will play a tone of given frequency, e.g: > > > > > > > This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms. > > Start avmd on the call with fs_cli: > avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start > > and inspect events by regestering to avmd events with: > events plain CUSTOM avmd::beep > > Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments. You can do this by subscription to TALK/NOTALK events. > Hope this helps. > > cheers, > Piotr > > On 4 February 2016 at 17:50, Piotr Gregor > wrote: > Hi, > "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. > Am I correct about that? " > > Yes. The AVMD module works by calculating the estimate of the frequency using DESA-2 algorithm. It fires amd::beep event when the variance for that estimate is below threshold. > The event is of form: > > Event-Subclass: avmd%3A%3Abeep > Event-Name: CUSTOM > Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 > FreeSWITCH-Hostname: home > FreeSWITCH-Switchname: home > FreeSWITCH-IPv4: 128.11.35.8 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2016-02-04%2016%3A56%3A55 > Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT > Event-Date-Timestamp: 1454605015915799 > Event-Calling-File: mod_avmd.c > Event-Calling-Function: avmd_process > Event-Calling-Line-Number: 556 > Event-Sequence: 950 > Beep-Status: stop > Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a > call-command: avmd > > The AVMD doesn't take into consideration TALK/NOTALK events. > > > "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? " > > You can test how it works by creating extension that will play a tone of given frequency, e.g: > > > > > > > This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms. > > Start avmd on the call with fs_cli: > avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start > > and inspect events by regestering to avmd events with: > events plain CUSTOM avmd::beep > > Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments. > Hope this helps. > > cheers, > Piotr > > > > > > > > On 3 February 2016 at 19:50, Christopher Rienzo > wrote: > I wouldn't use beep detection in a dialer application, but it could be useful in something like follow me to reduce the occurrence of voicemails being left on subscriber phones. > > On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig > wrote: > I have a client that needs to reliably detect if their outbound calls are answered by a human or voicemail system, so that they can take different actions based on that determination. > > I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. > > Am I correct about that? How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? > > Thanks! > Bob > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/05c384f9/attachment-0001.html From anthony.minessale at gmail.com Fri Feb 5 03:02:51 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Feb 2016 18:02:51 -0600 Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit Message-ID: This weekend marks the beginning of the 2016 FreeSWITCH summit where the team meets up to work on ClueCon and code and the next release of FreeSWITCH. Now is your chance to virtually buy the guys a Beer or help fund part of their meal! paypal at freeswitch.org http://bit.ly/1L1Azn9 -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/31443824/attachment.html From max at nysolutions.com Fri Feb 5 06:02:22 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 5 Feb 2016 03:02:22 +0000 Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit In-Reply-To: References: Message-ID: Link not working. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, February 4, 2016 7:03 PM To: Freeswitch-users Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit This weekend marks the beginning of the 2016 FreeSWITCH summit where the team meets up to work on ClueCon and code and the next release of FreeSWITCH. Now is your chance to virtually buy the guys a Beer or help fund part of their meal! paypal at freeswitch.org http://bit.ly/1L1Azn9 -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/6b1a867a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/6b1a867a/attachment.jpg From blake at cogents.io Fri Feb 5 06:40:27 2016 From: blake at cogents.io (Blake Priddy) Date: Thu, 4 Feb 2016 21:40:27 -0600 Subject: [Freeswitch-users] Fellow FreeSWITCHERS Message-ID: It's time for the freeswitch annual dev meeting. The freeswitch developers we love and care for so much will be gathering to talk business, future development, and just talking and networking! I am saying this because I have found out that most of the devs are paying out of their pocket for their dinner, hotel, and travel. This is not okay with me. Therefore I have made a sizeable donation on the freeswitch website for the developers to use for food, room, flight, etc. I honestly don't care what is done with my donation, I just know it will be put to good use. I am encouraging all of you out there to spare some funds for this team that has helped us make money, and/or help others save money. Let's give this team the meal they deserve! Thanks freeswitch devs for all you do and continue to do to help others save money and be successful in their VoIP implementations. We appreciate you so much!! Support the team that built the dream! PayPal for the devs: paypal at freeswitch.org or visit the website and click the donate button! https://freeswitch.org/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160204/3fd6a5ed/attachment-0001.html From anthony.minessale at gmail.com Fri Feb 5 09:24:57 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Feb 2016 00:24:57 -0600 Subject: [Freeswitch-users] Time for the Annual FreeSWITCH Developers Summit In-Reply-To: References: Message-ID: Oops http://bit.ly/20KH4DR The paypal button on the site works as well or just paypal at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/299cd679/attachment.html From andrew at cassidywebservices.co.uk Fri Feb 5 12:28:26 2016 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 5 Feb 2016 09:28:26 +0000 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: <007701d15f95$365ceb00$a316c100$@botecomm.com> References: <003c01d15f73$d07a5190$716ef4b0$@botecomm.com> <007701d15f95$365ceb00$a316c100$@botecomm.com> Message-ID: The "info" application dumps all channel variables On 4 February 2016 at 21:44, Bote Man wrote: > Here?s what works for me: > > > > > > > > This will display it in fs_cli at least. > > > > I?m not familiar with the syntax of what you posted, unless that?s an API > command being pumped into FS via ESL or something. > > > > Bote > > > > > > *From:* Tim King > *Sent:* Thursday, 04 February, 2016 12:52 > > *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable? > > > > I keep trying to log them but the log application is showing me the log > statement and not the variable itself. I know I have used this successfully > before, but even if I try logging the destination_number it does not log. > > > > I have tried ${destination_number} and [${destination_number}] and my log > output looks like this. > > > > Action log(Log TEST ========== ${destination_number} or > [${destination_number}] > > > > On Thu, Feb 4, 2016 at 12:45 PM, Bote Man wrote: > > I also recommend using the FreeSWITCH ?log? application to display what FS > thinks those variables contain. > > > > --- > > Bote > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > *From:* Tim King > *Sent:* Thursday, 04 February, 2016 12:13 > *Subject:* Re: [Freeswitch-users] X-Auth-IP Variable? > > Thank you for the reply. I tried this for matching to the ACL but it is > failing. > > expression="true" break="on-false"/> > > > > On Thu, Feb 4, 2016 at 11:44 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > > ${variable_sip_h_X-Auth-IP} > > > > On 4 February 2016 at 16:19, Tim King wrote: > > I am using proxy authentication in my setup and it is working. To do this > I have created an acl > > *autoload_configs/acl.conf.xml* > > > > > > > > > > > > > > *sip_profiles/external.xml* > > > > > > > > > > > > This is all working as desired. The problem is prior to adding the > opensips I was using the network_addr variable in my dialplan. > > > > > > expression="true" break="on-false"/> > > expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> > > data="dtmf_type=rfc2833"/> > > data="accountcode=customer1123"/> > > data="continue_on_fail=false"/> > > data="hangup_after_bridge=true"/> > > data="proxy_media=true"/> > > data="sofia/external/$2 at 8.7.6.5:5060"/> > > > > > > > > This of course no longer works because network_addr is always the address > of my proxy server. How can I get the address from the X-authip into the > dialplan? > > > > > > > > -- > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > Managing Director > > > > *T *03300 100 960 *F > *03300 100 961 > > *E *andrew at cassidywebservices.co.uk > > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/f5eb488a/attachment-0001.html From lists at telefaks.de Fri Feb 5 12:30:44 2016 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 05 Feb 2016 10:30:44 +0100 Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt In-Reply-To: <007c01d15f95$a588b220$f09a1660$@botecomm.com> References: <56B3C09A.60907@telefaks.de> <007c01d15f95$a588b220$f09a1660$@botecomm.com> Message-ID: <56B46BC4.1010607@telefaks.de> Hello Bote, thanks for your reply. Maybe my post was a bit misleading. In both cases I have a shell opened on the machine and try from there running fs_cli.exe. The difference is * 1st case: I open a local shell on the Windows machine (via RDP connection), run cygwin, get a unix shell and then start fs_cli. Works. * 2nd case: Form an external machine I ssh to the Windows machine (sshd server is running there with cygwin), have a unix shell and then start fs_cli. Does not work as described below. So both cases are connecting to localhost:8021. And I am wondering now, why this behaves differently. Why Cygwin? For me, this is much easier, because I am used to Linux tools (grep, tail, editors, mc, ...). And it's much better for documentation purposes, when this all is text based and you have a large terminal window. Best regards Peter On 02/04/16 22:47, Bote Man wrote: > You can only connect to port 8021 locally because > > > > Is set in autoload_configs/event_socket.conf.xml > > If you set that to value to 0.0.0.0 the whole world can connect to port > 8021, but the problem with that is that the whole world can connect. I > wouldn't change without careful consideration of the security impact. > > I have never used ssh to connect to Windows machine, I didn't even know that > it is possible so that is a new one to me. I just remote desktop (mstsc.exe) > into the target machine and run the necessary commands and utilities > directly on the box. > > Sorry I couldn't be more help. > > > --- > Bote > FreeSWITCH Docs Janitor > http://freeswitch.org/confluence > > > > >> -----Original Message----- >> From: Peter Steinbach >> Sent: Thursday, 04 February, 2016 16:20 >> Subject: [Freeswitch-users] fs_cli under Windows and cygwin, no prompt >> >> I have inherited a Freeswitch installation on a Windows server, which I >> remotely manage with cygwin and ssh. >> >> This worked more or less nicely so far. But today I cannot manage >> Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no >> change. >> I can run fs_cli.exe, and I see the Freeswitch messages, but I do >> __not__ get a prompt. When I try to enter characters, nothing happens. >> >> On the machine itself, when I open a console and start cygwin, >> fs_cli.exe works as expected with prompt. But not via ssh. But >> fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be >> fine in general. >> >> What I understand is, that fs_cli connects to Freeswitch via port 8021. >> So when seeing FS messages, I expect that the connection is already >> fine, right? >> >> I also tried to connect via telnet to port 8021. This works in the local >> cygwin shell (get an auth request), but not via remote ssh shell (telnet >> terminates without message). >> >> I am puzzled. Anybody has a clue, why I do not have a prompt and where >> to search? >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists (att) telefaks.de >> Internet: www.telefaks.de >> >> >> __________________________________________________________ >> _______________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/1e16e69c/attachment.html From steveayre at gmail.com Fri Feb 5 15:24:34 2016 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 5 Feb 2016 12:24:34 +0000 Subject: [Freeswitch-users] X-Auth-IP Variable? In-Reply-To: References: Message-ID: It'll be ${sip_h_X-Auth-IP} - the variable_ prefix added by the info app is only to differentiate variables from fields. On 4 February 2016 at 16:44, Andrew Cassidy wrote: > ${variable_sip_h_X-Auth-IP} > > On 4 February 2016 at 16:19, Tim King wrote: > >> I am using proxy authentication in my setup and it is working. To do this >> I have created an acl >> *autoload_configs/acl.conf.xml* >> >> >> >> >> >> >> *sip_profiles/external.xml* >> >> >> >> >> >> This is all working as desired. The problem is prior to adding the >> opensips I was using the network_addr variable in my dialplan. >> >> >> > expression="true" break="on-false"/> >> > expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> >> > data="dtmf_type=rfc2833"/> >> > data="accountcode=customer1123"/> >> > data="continue_on_fail=false"/> >> > data="hangup_after_bridge=true"/> >> > data="proxy_media=true"/> >> > data="sofia/external/$2 at 8.7.6.5:5060"/> >> >> >> >> This of course no longer works because network_addr is always the address >> of my proxy server. How can I get the address from the X-authip into the >> dialplan? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/d8baa3ad/attachment.html From italo at freeswitch.org Fri Feb 5 21:44:27 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 5 Feb 2016 15:44:27 -0300 Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter In-Reply-To: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com> References: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com> Message-ID: Hi Rodrigo, We don't keep these counters in memory. If you want to figure out right now how many calls were abandoned or answered you need to parse the cdrs from these calls or listen to the ESL events and keep a counter. But, adding realtime counters to mod_callcenter shouldn't be difficult, a PR would be awesome ;) On Wed, Feb 3, 2016 at 9:35 PM, Rodrigo Ram?rez Norambuena < decipher.hk at gmail.com> wrote: > Hello everyone!, > > I'm testing a mod_callcenter for make a FreeSWITCH version of a open > source software > (https://github.com/roramirez/qpanel/tree/fs) > > Now i using ESL to send command and get information from the module. I > looking for a way to know the > data of answered/abandoned call from a queue and agents. > > Somebody can give a tips or a light? > > Regards, > -- > Rodrigo Ram?rez Norambuena > http://www.rodrigoramirez.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/6110b39a/attachment-0001.html From italo at freeswitch.org Fri Feb 5 21:51:11 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 5 Feb 2016 15:51:11 -0300 Subject: [Freeswitch-users] mod_xml_curl In-Reply-To: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com> References: <037a01d15f8a$c24e1e00$46ea5a00$@imerchantsystems.com> Message-ID: Londre, {domain} should be ${domain} If mod_xml_curl is not respecting your configuration file then we have a bug, can you please file a JIRA with your configs and log files? On Thu, Feb 4, 2016 at 5:29 PM, Londre wrote: > Hi, > > > > I have read the wiki several times and followed all the steps to the best > of my knowledge. I have gotten mod_xml_curl build and making requests. I > set xml_curl debug_on and I can see the request that freeswitch sends and I > can take a look at the file it generates in the /tmp/ directory. However > the file doesn?t seem to be writing anything. I have my web service set up > to hand a POST request but it looks like freeswitch is sending GET requests > even though I set the method to POST in xml_curl.xml. Any ideal would could > be going wrong? > > > > System Info: > > Debian Jessie > > Freeswitch 1.7 built from source. > > > > Xml_curl.xml: > > > > 1 > > 2 > > 3 > > 4 bindings="directory|dialplan"/> > > 5 > > 6 > > 7 > > 8 > > 9 > > > > It also doesn?t seem to be sending the variables for the > gateway-credentials. Its seems weird that its sending a GET request when > the default should be a POST. Also it is a little confusing on the wiki > some things show a query string and other show header variables. I?d prefer > to use POST. > > > > Please offer some insight into what could be going wrong. > > > > *Londre Blocker* > > Developer > > iMerchant Systems Inc. > > T: 844.727.1998 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/869ba051/attachment.html From nneul at mst.edu Sat Feb 6 00:45:28 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 5 Feb 2016 15:45:28 -0600 Subject: [Freeswitch-users] FYI in case anyone else was using the CallCap blocklist Message-ID: <56B517F8.8070200@mst.edu> It's been taken offline by the company, and they indicate they have no plans to bring the XML download back. Is anyone here making any use of a similar data feed, or the FTC robocall complaint data or similar? Any recommendations? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From peter at hartmanncomputer.com Sat Feb 6 01:01:40 2016 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Fri, 5 Feb 2016 17:01:40 -0500 Subject: [Freeswitch-users] mod_esf with polycom In-Reply-To: <003f01d15b95$07d0e380$1772aa80$@botecomm.com> References: <008001d1598c$ef0aa200$cd1fe600$@botecomm.com> <003f01d15b95$07d0e380$1772aa80$@botecomm.com> Message-ID: > What do you mean by ?group?? It's a Polycom thing...check out the pdf Brian referenced. Channel is also a polycom thing. http://support.polycom.com/global/documents/support/technical/products/voice/Audio_Packet_Format.pdf I got it going.....it turns our that Paging Group 25 is channel 50 at the packet level. Thank you, Brian! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it. On Sat, Jan 30, 2016 at 2:32 PM, Bote Man wrote: > > At this point I would try specifying all fields whether they are needed or not. Try > > > > Where 4 is the TTL; I?m just guessing that maybe these packets must hop through a router or two. The 6061 does not apply, but must be the third argument to get to the fourth argument. > > > > What do you mean by ?group?? The arguments specified on the Confluence page are the only ones that control the multicast page application. If Polycom needs additional processing then this app won?t work for you. I have never used Polycom so I can?t say, maybe others on here who have used Polycom could chime in. > > > > FYI, the multicast stream is sent out via UDP. > > > > > > --- > > Bote > > > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > > > > > From: Peter Hartmann > Sent: Friday, 29 January, 2016 14:41 > > > Subject: Re: [Freeswitch-users] mod_esf with polycom > > > > I just did a packet capture of a mod_esf call. What I'm seeing is not a lot compared to a polycom page capture. In the mod_esf cap, I just a handful of packets of protocol IGMPv3. Regardless of my config: > > > > They are being sent to the wrong destination address 224.0.0.22. And curiously, I see my destination address being used as the group. (see attachment) > > > > I don't see this 224.0.0.22 address in the source for mod_esf at all....I wonder where that's coming from? > > > > In the polycom capture, there are no such IGMP packets, just UDP. Although I do realize that it's supposed to do a few different broadcast types from different manufacturers all at once. > > > > > > Peter Hartmann > Hartmann Computer Consulting > http://blog.hartmanncomputer.com > (212)203-8870 > > If I can't explain it to you in plain language, that means I don't understand it. > > > > > > On Thu, Jan 28, 2016 at 12:30 AM, Bote Man wrote: > > Perhaps you are being led astray by the term ?multicast group? in the usage prototype. That is simply the multicast address, such as 239.1.1.1 > > > > FIXED! > > > > IGMP = Internet Group Management Protocol which is where that came from. > > > > I have used mod_esf to send multicast streams to devices manufactured by one of my clients and it worked swimmingly with only the multicast address and port being the fields that mattered. Thanks to a change by anthm you can even add it to the conference bridge if you want both regular SIP calls and pages to be combined into one group announcement. > > > > With a phone there?s no telling what else they might require to light it up for a multicast announcement. Somebody was just asking a very similar question here last week for a different phone. If the phone requires some control signal or command to open up the speaker for a multicast announcement to follow, then I don?t think FS does that right now. It might be worth scouring Polycom documentation to see if they reveal any secrets. > > > > * * * > > I really wish the order of the arguments were > > Address, port, TTL, [other stuff] > > So that ?other stuff? could?ve been made optional. I have never gotten a clear definition of what that Linksys control port does, but my research revealed that it was likely carried over from an Asterisk compatibility feature. > > > > > > --- > > Bote > > > > FreeSWITCH Docs Janitor > > http://freeswitch.org/confluence > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Hartmann > Sent: Wednesday, 27 January, 2016 22:27 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_esf with polycom > > > > Thanks Brian. Yes, I've tried it. It looks promising though, since I can see some of the default polycom settings defined in variables. Nothing about group is mentioned...so maybe it's going to the default group? I have a feeling I'll be doing a packet capture on a group-25 (emergency) broadcast to see what it does. > > > > > > Peter Hartmann > Hartmann Computer Consulting > http://blog.hartmanncomputer.com > (212)203-8870 > > If I can't explain it to you in plain language, that means I don't understand it. > > > > On Wed, Jan 27, 2016 at 10:03 PM, Brian West wrote: > > I'll review the source in the morning and verify if it's documented correctly. > > > > On Wednesday, January 27, 2016, Brian West wrote: > > Have you tried it? It should just work if I recall correctly. > > On Wednesday, January 27, 2016, Peter Hartmann wrote: > > Hey, how do you use mod_esf with Polycom? The source suggests that > it's supported, but the arguments in the documentation seem specific > to Linksys. From using the feature directly from a handset, I imagine > we have to pass multicast-address, multicast-port and group-number. > I've tried every combination and permutation with no luck. > > The reason I'm doing this is because if the emergency group number is > used, the announcements will be at full volume regardless of the > hands-free volume setting. > > And also I'm looking to do this in FS because I'm bridging in another > PA system also. > > > Thanks, > > > Peter Hartmann > Hartmann Computer Consulting > http://blog.hartmanncomputer.com > (212)203-8870 > > If I can't explain it to you in plain language, that means I don't > understand it. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here! | Reddit: /r/freeswitch > > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > > > > -- > > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here! | Reddit: /r/freeswitch > > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Sat Feb 6 03:37:33 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Feb 2016 18:37:33 -0600 Subject: [Freeswitch-users] Fellow FreeSWITCHERS In-Reply-To: References: Message-ID: Thanks so much Blake! On Thu, Feb 4, 2016 at 9:40 PM, Blake Priddy wrote: > It's time for the freeswitch annual dev meeting. The freeswitch developers > we love and care for so much will be gathering to talk business, future > development, and just talking and networking! I am saying this because I > have found out that most of the devs are paying out of their pocket for > their dinner, hotel, and travel. This is not okay with me. Therefore I have > made a sizeable donation on the freeswitch website for the developers to > use for food, room, flight, etc. I honestly don't care what is done with my > donation, I just know it will be put to good use. I am encouraging all of > you out there to spare some funds for this team that has helped us make > money, and/or help others save money. Let's give this team the meal they > deserve! Thanks freeswitch devs for all you do and continue to do to help > others save money and be successful in their VoIP implementations. We > appreciate you so much!! > > Support the team that built the dream! > > PayPal for the devs: paypal at freeswitch.org > > or visit the website and click the donate button! https://freeswitch.org/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160205/94c36c5a/attachment-0001.html From deepikay at iiitd.ac.in Sat Feb 6 17:24:02 2016 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Sat, 6 Feb 2016 19:54:02 +0530 Subject: [Freeswitch-users] Suspicious Incoming Calls Message-ID: Hi, I have microsip installed in my windows configured for one or two SIP accounts for different Freeswitch servers. I am receiving a call from 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, these Freeswitch servers are hosted at cloud machine. Is it a case of hacking the servers. What measures should I take to secure both my servers and system. Regards, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/3e9ed5e7/attachment.html From brian at freeswitch.org Sat Feb 6 17:28:33 2016 From: brian at freeswitch.org (Brian West) Date: Sat, 6 Feb 2016 08:28:33 -0600 Subject: [Freeswitch-users] Fellow FreeSWITCHERS In-Reply-To: References: Message-ID: Thank you! You rock! On Thursday, February 4, 2016, Blake Priddy wrote: > It's time for the freeswitch annual dev meeting. The freeswitch developers > we love and care for so much will be gathering to talk business, future > development, and just talking and networking! I am saying this because I > have found out that most of the devs are paying out of their pocket for > their dinner, hotel, and travel. This is not okay with me. Therefore I have > made a sizeable donation on the freeswitch website for the developers to > use for food, room, flight, etc. I honestly don't care what is done with my > donation, I just know it will be put to good use. I am encouraging all of > you out there to spare some funds for this team that has helped us make > money, and/or help others save money. Let's give this team the meal they > deserve! Thanks freeswitch devs for all you do and continue to do to help > others save money and be successful in their VoIP implementations. We > appreciate you so much!! > > Support the team that built the dream! > > PayPal for the devs: paypal at freeswitch.org > > > or visit the website and click the donate button! https://freeswitch.org/ > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/61021f67/attachment.html From jude19love at gmail.com Sat Feb 6 18:50:23 2016 From: jude19love at gmail.com (Jude Mukundane) Date: Sat, 6 Feb 2016 15:50:23 +0000 Subject: [Freeswitch-users] Suspicious Incoming Calls In-Reply-To: References: Message-ID: Hello Deepika, This is common for anyone running FS in the cloud with discoverable ports. Lots of people just run scripts that crawl the internet in search of SIP servers. After getting one they try bogus invites to try and see if they can get calls through - if your server is forwarding to PSTN, you could end up with a bill in thousands of dollars in minutest. In my case, I use a simple IP Table in Ubuntu (more like an access control list) to define allowed and non allowed IPs. Can someone please elaborate on a measure that inolves config level security because blocking out masses is not goot Internet Citizenry. Jude On Sat, Feb 6, 2016 at 2:24 PM, Deepika Yadav wrote: > Hi, > > I have microsip installed in my windows configured for one or two SIP > accounts for different Freeswitch servers. I am receiving a call from > 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, > these Freeswitch servers are hosted at cloud machine. > > Is it a case of hacking the servers. What measures should I take to secure > both my servers and system. > > Regards, > Deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/2d315a09/attachment.html From aqsyounas at gmail.com Sat Feb 6 19:01:03 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 6 Feb 2016 21:01:03 +0500 Subject: [Freeswitch-users] Forward call to a destination without changing RURI Message-ID: Hi, I receive Invite from vendor with i need to forward to a destination ip set in special header X-Dest without changing the RURI. When I try to bridge this invite It changes the RURI. I need to preserve the RURI but set destination to X-Dest header. How can i forward call to that IP without changing the RURI. Best Regard. This email has been sent from a virus-free computer protected by Avast. www.avast.com <#396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/bceb8e35/attachment.html From krice at freeswitch.org Sat Feb 6 19:36:12 2016 From: krice at freeswitch.org (Ken Rice) Date: Sat, 6 Feb 2016 10:36:12 -0600 Subject: [Freeswitch-users] Suspicious Incoming Calls In-Reply-To: References: Message-ID: <663601d160fc$7e239020$7a6ab060$@freeswitch.org> http://youtu.be/oor-liSVL0o From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deepika Yadav Sent: Saturday, February 6, 2016 8:24 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Suspicious Incoming Calls Hi, I have microsip installed in my windows configured for one or two SIP accounts for different Freeswitch servers. I am receiving a call from 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, these Freeswitch servers are hosted at cloud machine. Is it a case of hacking the servers. What measures should I take to secure both my servers and system. Regards, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/62c622c2/attachment-0001.html From s.safarov at gmail.com Sat Feb 6 20:07:28 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 6 Feb 2016 20:07:28 +0300 Subject: [Freeswitch-users] Forward call to a destination without changing RURI In-Reply-To: References: Message-ID: Try set sip_req_uri variable. Sergey On Sat, Feb 6, 2016 at 7:01 PM, Aqs Younas wrote: > Hi, > > I receive Invite from vendor with i need to forward to a destination ip > set in special header X-Dest without changing the RURI. > > When I try to bridge this invite > > > > It changes the RURI. I need to preserve the RURI but set destination to > X-Dest header. > > How can i forward call to that IP without changing the RURI. > > > Best Regard. > This email has been sent from a > virus-free computer protected by Avast. > www.avast.com > <#-1084937732_396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/9f02fd6a/attachment.html From ahabiba at gmail.com Sat Feb 6 20:10:42 2016 From: ahabiba at gmail.com (Ahmed Habiba) Date: Sat, 6 Feb 2016 20:10:42 +0300 Subject: [Freeswitch-users] Suspicious Incoming Calls In-Reply-To: References: Message-ID: Hello, You have to use one of the below options: Option1: if you are not allowing another system to access you system without username and password i.e. you make your system as sip gateway for other trusted company, the you can remove the file named ?external.xml? under /usr/local/freeswitch/conf/sip_profiles/, then either restart your instance or run ?reload mod_sofia? in fs_cli Note be sure that you take a copy of external.xml before you remove it. Option2: add the below line in you external.xml profile mentioned above, this will not allow any external system to login expect if it has been allowed in you ACL list or it has a username/password this will make things little hard, then you may install fail2ban module. all cases you need to restart your profiles. Thanks, Ahmed Habiba > > > > From: Jude Mukundane > Subject: Re: [Freeswitch-users] Suspicious Incoming Calls > Date: February 6, 2016 at 6:50:23 PM GMT+3 > To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > > > Hello Deepika, > > This is common for anyone running FS in the cloud with discoverable ports. Lots of people just run scripts that crawl the internet in search of SIP servers. After getting one they try bogus invites to try and see if they can get calls through - if your server is forwarding to PSTN, you could end up with a bill in thousands of dollars in minutest. In my case, I use a simple IP Table in Ubuntu (more like an access control list) to define allowed and non allowed IPs. > > Can someone please elaborate on a measure that inolves config level security because blocking out masses is not goot Internet Citizenry. > > Jude > > On Sat, Feb 6, 2016 at 2:24 PM, Deepika Yadav > wrote: > Hi, > > I have microsip installed in my windows configured for one or two SIP accounts for different Freeswitch servers. I am receiving a call from 2022 at myPublicIP repeatitively even if I disconnect from all the accounts, these Freeswitch servers are hosted at cloud machine. > > Is it a case of hacking the servers. What measures should I take to secure both my servers and system. > > Regards, > Deepika > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/248d2206/attachment.html From aqsyounas at gmail.com Sat Feb 6 20:52:42 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 6 Feb 2016 22:52:42 +0500 Subject: [Freeswitch-users] Forward call to a destination without changing RURI In-Reply-To: References: Message-ID: Thanks for your replay. Setting sip_req_uri variable was giving me internal server error. fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:40 sofia/external/14703999454 at 45.56.70.29:5070 Standard INIT fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:48 (sofia/external/14703999454 at 45.56.70.29:5070) State Change CS_INIT -> CS_ROUTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:516 (sofia/external/14703999454 at 45.56.70.29:5070) State INIT going to sleep fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070) Running State Change CS_ROUTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] sofia.c:6760 Channel sofia/external/14703999454 at 45.56.70.29:5070 entering state [terminated][900] fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [NOTICE] sofia.c:7779 Hangup sofia/external/14703999454 at 45.56.70.29:5070 [CS_ROUTING] [NORMAL_UNSPECIFIED] fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:532 (sofia/external/14703999454 at 45.56.70.29:5070) State ROUTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:141 sofia/external/14703999454 at 45.56.70.29:5070 SOFIA ROUTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:532 (sofia/external/14703999454 at 45.56.70.29:5070) State ROUTING going to sleep fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070) Running State Change CS_HANGUP fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:739 (sofia/external/14703999454 at 45.56.70.29:5070) Callstate Change DOWN -> HANGUP fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:741 (sofia/external/14703999454 at 45.56.70.29:5070) State HANGUP fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:431 Channel sofia/external/14703999454 at 45.56.70.29:5070 hanging up, cause: NORMAL_UNSPECIFIED fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:60 sofia/external/14703999454 at 45.56.70.29:5070 Standard HANGUP, cause: NORMAL_UNSPECIFIED fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:741 (sofia/external/14703999454 at 45.56.70.29:5070) State HANGUP going to sleep fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:508 (sofia/external/14703999454 at 45.56.70.29:5070) State Change CS_HANGUP -> CS_REPORTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:473 (sofia/external/14703999454 at 45.56.70.29:5070) Running State Change CS_REPORTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:827 (sofia/external/14703999454 at 45.56.70.29:5070) State REPORTING fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:104 sofia/external/14703999454 at 45.56.70.29:5070 Standard REPORTING, cause: NORMAL_UNSPECIFIED fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:827 (sofia/external/14703999454 at 45.56.70.29:5070) State REPORTING going to sleep 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_ivr_originate.c:3751 Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED] fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:499 (sofia/external/14703999454 at 45.56.70.29:5070) State Change CS_REPORTING -> CS_DESTROY fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_session.c:1646 Session 50 (sofia/external/ 14703999454 at 45.56.70.29:5070) Locked, Waiting on external entities fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [NOTICE] switch_core_session.c:1664 Session 50 (sofia/external/ 14703999454 at 45.56.70.29:5070) Ended fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/ 14703999454 at 45.56.70.29:5070 [CS_DESTROY] fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:630 (sofia/external/14703999454 at 45.56.70.29:5070) Running State Change CS_DESTROY 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [INFO] mod_dptools.c:3379 Originate Failed. Cause: NORMAL_UNSPECIFIED 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [NOTICE] switch_channel.c:4804 Hangup sofia/external/sipp at 104.237.140.13:5061 [CS_EXECUTE] [NORMAL_UNSPECIFIED] fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:640 (sofia/external/14703999454 at 45.56.70.29:5070) State DESTROY fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:341 sofia/external/14703999454 at 45.56.70.29:5070 SOFIA DESTROY fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:111 sofia/external/14703999454 at 45.56.70.29:5070 Standard DESTROY fd3863cc-ecd6-481f-80c6-102fb0d31fbe 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:640 (sofia/external/14703999454 at 45.56.70.29:5070) State DESTROY going to sleep 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_session.c:2796 sofia/external/sipp at 104.237.140.13:5061 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:539 (sofia/external/sipp at 104.237.140.13:5061) State EXECUTE going to sleep 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:473 (sofia/external/sipp at 104.237.140.13:5061) Running State Change CS_HANGUP 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:739 (sofia/external/sipp at 104.237.140.13:5061) Callstate Change RINGING -> HANGUP 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 104.237.140.13:5061) State HANGUP 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:425 sofia/external/sipp at 104.237.140.13:5061 Overriding SIP cause 480 with 900 from the other leg 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:431 Channel sofia/external/sipp at 104.237.140.13:5061 hanging up, cause: NORMAL_UNSPECIFIED 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 900 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 104.237.140.13:5061 Standard HANGUP, cause: NORMAL_UNSPECIFIED 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 104.237.140.13:5061) State HANGUP going to sleep 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:508 (sofia/external/sipp at 104.237.140.13:5061) State Change CS_HANGUP -> CS_REPORTING 9f034ddd-6548-449c-8299-408d0d5cba9b 2016-02-06 17:23:18.139328 [DEBUG] switch_core_state_machine.c:473 (sofia/external/sipp at 104.237.140.13:5061) Running State Change CS_REPORTING send 296 bytes to udp/[104.237.140.13]:5061 at 17:23:18.156113: But got solution by using fs_path. On 6 February 2016 at 22:07, Sergey Safarov wrote: > Try set sip_req_uri variable. > > Sergey > > On Sat, Feb 6, 2016 at 7:01 PM, Aqs Younas wrote: > >> Hi, >> >> I receive Invite from vendor with i need to forward to a destination ip >> set in special header X-Dest without changing the RURI. >> >> When I try to bridge this invite >> >> >> >> It changes the RURI. I need to preserve the RURI but set destination to >> X-Dest header. >> >> How can i forward call to that IP without changing the RURI. >> >> >> Best Regard. >> This email has been sent from a >> virus-free computer protected by Avast. >> www.avast.com >> <#1808395836_-1084937732_396301616_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/11067c80/attachment-0001.html From koralu at gmail.com Sat Feb 6 21:29:39 2016 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 6 Feb 2016 20:29:39 +0200 Subject: [Freeswitch-users] Run php script on execute_on_answer Message-ID: Hello, I try to execute a php script after the call is answered but I can't figure how. I need something like this. So what I try to do is to pass the uuid and caller-destination-number to an external source after the call is answered. Is any other approach? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/e3eb561b/attachment.html From krice at freeswitch.org Sat Feb 6 21:55:15 2016 From: krice at freeswitch.org (Ken Rice) Date: Sat, 6 Feb 2016 12:55:15 -0600 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: Message-ID: You cant run an http location you need to call it with curl Sent from my iPhone > On Feb 6, 2016, at 12:29 PM, Adrian Andrei wrote: > > Hello, > > I try to execute a php script after the call is answered but I can't figure how. I need something like this. > > > > > > > > So what I try to do is to pass the uuid and caller-destination-number to an external source after the call is answered. Is any other approach? > > > Thank you > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/7e9fa90c/attachment.html From koralu at gmail.com Sat Feb 6 23:03:59 2016 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 6 Feb 2016 22:03:59 +0200 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: Message-ID: Ok. It works on a separate extension, but I can't figure out what is the syntax for inline approach This doesn't work. On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice wrote: > You cant run an http location you need to call it with curl > > Sent from my iPhone > > On Feb 6, 2016, at 12:29 PM, Adrian Andrei wrote: > > Hello, > > I try to execute a php script after the call is answered but I can't > figure how. I need something like this. > > > > > > > > So what I try to do is to pass the uuid and caller-destination-number to > an external source after the call is answered. Is any other approach? > > > Thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/596958fd/attachment.html From denis.papes at zg.t-com.hr Sat Feb 6 23:10:36 2016 From: denis.papes at zg.t-com.hr (=?UTF-8?Q?Denis_Pape=c5=a1?=) Date: Sat, 6 Feb 2016 21:10:36 +0100 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: Message-ID: <56B6533C.2090105@zg.t-com.hr> You should load mod_curl and use syntax On 02/06/2016 09:03 PM, Adrian Andrei wrote: > Ok. It works on a separate extension, but I can't figure out what is > the syntax for inline approach > > > > This doesn't work. > > > > On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice > wrote: > > You cant run an http location you need to call it with curl > > Sent from my iPhone > > On Feb 6, 2016, at 12:29 PM, Adrian Andrei > wrote: > >> Hello, >> >> I try to execute a php script after the call is answered but I >> can't figure how. I need something like this. >> >> >> >> >> >> >> >> So what I try to do is to pass the uuid and >> caller-destination-number to an external source after the call is >> answered. Is any other approach? >> >> >> Thank you >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/fc4a470e/attachment-0001.html From koralu at gmail.com Sat Feb 6 23:40:07 2016 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 6 Feb 2016 22:40:07 +0200 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: <56B6533C.2090105@zg.t-com.hr> References: <56B6533C.2090105@zg.t-com.hr> Message-ID: Mod_curl is loaded and in separate extension it's working fine. The problem is with the following syntax: I need something like this: On Sat, Feb 6, 2016 at 10:10 PM, Denis Pape? wrote: > You should load mod_curl and use syntax > > > > > > On 02/06/2016 09:03 PM, Adrian Andrei wrote: > > Ok. It works on a separate extension, but I can't figure out what is the > syntax for inline approach > > > > This doesn't work. > > > > On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice wrote: > >> You cant run an http location you need to call it with curl >> >> Sent from my iPhone >> >> On Feb 6, 2016, at 12:29 PM, Adrian Andrei < >> koralu at gmail.com> wrote: >> >> Hello, >> >> I try to execute a php script after the call is answered but I can't >> figure how. I need something like this. >> >> >> >> >> >> >> >> So what I try to do is to pass the uuid and caller-destination-number to >> an external source after the call is answered. Is any other approach? >> >> >> Thank you >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.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-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/c7654b9c/attachment.html From denis.papes at zg.t-com.hr Sat Feb 6 23:45:41 2016 From: denis.papes at zg.t-com.hr (=?UTF-8?Q?Denis_Pape=c5=a1?=) Date: Sat, 6 Feb 2016 21:45:41 +0100 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: <56B6533C.2090105@zg.t-com.hr> Message-ID: <56B65B75.9050402@zg.t-com.hr> Sorry, I just copy/pasted your previous line without looking. It should be On 02/06/2016 09:40 PM, Adrian Andrei wrote: > Mod_curl is loaded and in separate extension it's working fine. The > problem is with the following syntax: > > I need something like this: > > > > > On Sat, Feb 6, 2016 at 10:10 PM, Denis Pape? > wrote: > > You should load mod_curl and use syntax > > > > > > On 02/06/2016 09:03 PM, Adrian Andrei wrote: >> Ok. It works on a separate extension, but I can't figure out what >> is the syntax for inline approach >> >> >> >> This doesn't work. >> >> >> >> On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice > > wrote: >> >> You cant run an http location you need to call it with curl >> >> Sent from my iPhone >> >> On Feb 6, 2016, at 12:29 PM, Adrian Andrei > > wrote: >> >>> Hello, >>> >>> I try to execute a php script after the call is answered but >>> I can't figure how. I need something like this. >>> >>> >>> >>> >>> >>> >>> >>> So what I try to do is to pass the uuid and >>> caller-destination-number to an external source after the >>> call is answered. Is any other approach? >>> >>> >>> Thank you >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/419c11c2/attachment-0001.html From krice at freeswitch.org Sat Feb 6 23:56:28 2016 From: krice at freeswitch.org (Ken Rice) Date: Sat, 6 Feb 2016 14:56:28 -0600 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: Message-ID: <5DFC87DE-7046-4EAF-A9A6-73506BBD9594@freeswitch.org> Why call system there? See mod curl thats what its for Sent from my iPhone > On Feb 6, 2016, at 2:03 PM, Adrian Andrei wrote: > > Ok. It works on a separate extension, but I can't figure out what is the syntax for inline approach > > > > This doesn't work. > > > >> On Sat, Feb 6, 2016 at 8:55 PM, Ken Rice wrote: >> You cant run an http location you need to call it with curl >> >> Sent from my iPhone >> >>> On Feb 6, 2016, at 12:29 PM, Adrian Andrei wrote: >>> >>> Hello, >>> >>> I try to execute a php script after the call is answered but I can't figure how. I need something like this. >>> >>> >>> >>> >>> >>> >>> >>> So what I try to do is to pass the uuid and caller-destination-number to an external source after the call is answered. Is any other approach? >>> >>> >>> Thank you >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/28cc335e/attachment.html From anton.vojlenko at gmail.com Sun Feb 7 00:38:34 2016 From: anton.vojlenko at gmail.com (Anton) Date: Sat, 6 Feb 2016 23:38:34 +0200 Subject: [Freeswitch-users] WebSocket behind NGINX In-Reply-To: <56AD0CE7.6000607@gmail.com> References: <56AD0CE7.6000607@gmail.com> Message-ID: <56B667DA.4010505@gmail.com> Hi, Sorry for not answering for a long time. Dan, thank you, your recommendation really helped me. So in order to proxy websocket request you need: 1. Proxy websocket requests in this way WSS -> (NGINX) -> FS WSS or WS -> (NGINX) -> FS WS 2. Modify local-network-acl 3. Modify apply-candidate-acl if you would like to drop more rtp candidates PS: I highly recommend to watch this video about NAT issues and ACL configuration: https://www.youtube.com/watch?v=_WSx-T6TriI BR, Anton Voylenko On 01/30/2016 09:20 PM, Anton wrote: > Hello All, > > I have to proxy all websocket requests though a nginx server. Right > now I am using next configuration: > > map $http_upgrade $connection_upgrade { > default upgrade; > '' close; > } > > server { > listen 443; > server_name wss.somedomain.com.ua; > > ssl on; > ssl_certificate /etc/nginx/cert.pem; > ssl_certificate_key /etc/nginx/private.key; > > location / { > proxy_pass http://127.0.0.1:5066; > proxy_http_version 1.1; > proxy_set_header Upgrade $http_upgrade; > proxy_set_header Connection $connection_upgrade; > proxy_read_timeout 86400s; > } > > access_log /var/log/nginx/wss_access; > error_log /var/log/nginx/wss_error debug; > } > > I dumped traffic from nginx and found out that "switching protocol" > phrase was successful but INVITE message from my browser in pending > state. > Maybe FreeSWITCH wants real IP not loopback? Who have faced with > similar problem? > > BR, > Anton From william.suffill at gmail.com Sun Feb 7 01:02:40 2016 From: william.suffill at gmail.com (William Suffill) Date: Sat, 6 Feb 2016 17:02:40 -0500 Subject: [Freeswitch-users] FYI in case anyone else was using the CallCap blocklist In-Reply-To: <56B517F8.8070200@mst.edu> References: <56B517F8.8070200@mst.edu> Message-ID: I'm also curious to what the community uses for this. I been using TrueSpam scores from TrueCNAM and it's be ok on my small personal usage [Free Tier]. I also have been manually blocking callerids so repeated calls get dropped automatically. On Fri, Feb 5, 2016 at 4:45 PM, Nathan Neulinger wrote: > It's been taken offline by the company, and they indicate they have no > plans to bring the XML download back. > > Is anyone here making any use of a similar data feed, or the FTC robocall > complaint data or similar? Any recommendations? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/5ead7f45/attachment.html From gregor at infomedia.si Sat Feb 6 21:36:39 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 06 Feb 2016 18:36:39 +0000 Subject: [Freeswitch-users] Run php script on execute_on_answer In-Reply-To: References: Message-ID: Try to use https://wiki.freeswitch.org/wiki/Mod_curl And execute it as api on answer. On Sat, Feb 6, 2016, 19:31 Adrian Andrei wrote: > Hello, > > I try to execute a php script after the call is answered but I can't > figure how. I need something like this. > > > > > > > > So what I try to do is to pass the uuid and caller-destination-number to > an external source after the call is answered. Is any other approach? > > > Thank you > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160206/7590399a/attachment-0001.html From gregor at infomedia.si Sun Feb 7 08:51:08 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 7 Feb 2016 06:51:08 +0100 Subject: [Freeswitch-users] TCP registrations In-Reply-To: References: <001b01d15d51$455900d0$d00b0270$@botecomm.com> <56B0ACD2.9070800@wirelessmundi.com> Message-ID: Everything works as expected :-) ALG on our router caused this problem. It looks that ALG blocked outgoing SIP packages via TCP. Thank you for your time. 2016-02-02 18:07 GMT+01:00 Brian West : > I'm going to guess your device probably fails to send the transport=tcp on > the contact there for it probably registers over TCP but we contact it back > over UDP? Can you confirm? > > On Tue, Feb 2, 2016 at 7:19 AM, Antonio Silva > wrote: > >> The parameter is "bind-params" >> >> >> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files >> >> by default fs should bind to tcp and udp but if you want only tcp just >> set for the profile: >> >> >> >> >> >> >> On 02/02/2016 02:01 PM, Sergey Safarov wrote: >> >> FS is defailt support UDP, TCP tansport. If you enable TLS in vars.xml >> also TLS transport is will be enabled. >> To check what is type of socket is open on server please use >> netstat -an --inet | grep -P "5060|5061|5080" >> >> Example >> [admin at node1.sbc ~]$ netstat -an --inet | grep -P "5060|5061|5080" >> tcp 0 0 217.12.247.214:5060 0.0.0.0:* >> LISTEN >> tcp 0 0 10.21.7.30:5060 >> >> 0.0.0.0:* LISTEN >> tcp 0 0 217.12.247.214:5061 0.0.0.0:* >> LISTEN >> tcp 0 0 217.12.247.214:5080 0.0.0.0:* >> LISTEN >> udp 0 0 217.12.247.214:5060 0.0.0.0:* >> >> udp 0 0 10.21.7.30:5060 >> >> 0.0.0.0:* >> udp 0 0 217.12.247.214:5080 0.0.0.0:* >> >> On Tue, Feb 2, 2016 at 11:50 AM, Gregor Nanger >> wrote: >> >>> Yes, I also think so, but cannot find explicitly documented. So please, >>> if anyone know exactly which command is, please help. >>> >>> 2016-02-02 1:32 GMT+01:00 Bote Man < >>> bote_radio at botecomm.com>: >>> >>>> FreeSWITCH uses UDP by default for SIP signaling. You can change this >>>> in the >>>> SIP_profile I believe. >>>> >>>> >>>> --- >>>> Bote >>>> >>>> FreeSWITCH Docs Janitor >>>> http://freeswitch.org/confluence >>>> >>>> >>>> >>>> >>>> > -----Original Message----- >>>> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch- >>>> > users-bounces at lists.freeswitch.org] On Behalf Of Gregor >>>> > Sent: Monday, 01 February, 2016 17:37 >>>> > To: freeswitch-users at lists.freeswitch.org >>>> > Subject: [Freeswitch-users] TCP registrations >>>> > >>>> > I think I am missing something. >>>> > >>>> > I would like to configure freeswitch that listens on TCP port for >>>> client >>>> > registrations (internal profile). As I read, freeswitch should do >>>> this by >>>> > default. But freeswitch responses only on UDP protocol. Is there a >>>> conf >>>> > setting for specify also tcp for registrations. >>>> > >>>> > >>>> > __________________________________________________________ >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Gregor Nanger >>> >>> *CTO* >>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 >>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia >>> ? www.infomedia.si >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.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-usershttp://www.freeswitch.org >> >> >> -- >> >> Saludos / Regards / Cumprimentos, >> Ant?nio silva >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here > ! > | Reddit: /r/freeswitch > > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia ? www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160207/735b3b05/attachment-0001.html From vbvbrj at gmail.com Sun Feb 7 17:59:42 2016 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 7 Feb 2016 16:59:42 +0200 Subject: [Freeswitch-users] hook to event or startup script which will hook to events? Message-ID: <56B75BDE.6020604@gmail.com> Hello. I'm running FS without problems for 3 years. I installed back then FS version 1.3.13b cad607d72e and it is running till now. I know it is old, but its stable. Back then I've used a start-up script in lua to catch callcenter's events and populate DB. Script is started and hooked to callcenter events. Now I want to add a event parse script for RECORD_STOP event. I've seen in lua.conf that I can use to autohook and execute script for every RECORD_STOP event. For testing I've write a small script to just log info: scriptname=argv[0] freeswitch.consoleLog("notice", scriptname.." Starting...\n") isdebug=false min_rec_secs=0 -- In seconds. api = freeswitch.API() recordings_dir = api:execute("global_getvar", "recordings_dir") databasename = api:execute("global_getvar", "freeswitch_data_db") databaseuser = api:execute("global_getvar", "freeswitch_databaseuser") databasepass = api:execute("global_getvar", "freeswitch_databasepass") function printlog (a,...) -- First paramter - line number -- Second parameter - lvl=debug,info,notice,warning,err,crit,alert -- Third parameter - the message -- Calling: printlog([line number,][log lvl,]message) local arg={...} line=0 lvl="debug" msg=a if (type(a)=='number' and a>0) then line=a else line=0 end maybe_lvl=string.lower(a) if (maybe_lvl=="debug" or maybe_lvl=="info" or maybe_lvl=="notice" or maybe_lvl=="warning" or maybe_lvl=="err" or maybe_lvl=="crit" or maybe_lvl=="alert") then lvl=maybe_lvl elseif (#arg>=1) then maybe_lvl=string.lower(arg[1]) if (maybe_lvl=="debug" or maybe_lvl=="info" or maybe_lvl=="notice" or maybe_lvl=="warning" or maybe_lvl=="err" or maybe_lvl=="crit" or maybe_lvl=="alert") then lvl=maybe_lvl end end if (#arg==1) then msg=arg[1] end if (#arg==2) then msg=arg[2] end if not (isdebug and lvl=="debug") then freeswitch.consoleLog(lvl,scriptname .. " (" .. line .."): " .. msg .. "\n") end end--]] freeswitch.consoleLog("notice", scriptname..event:serialize("").."\n") After restarint FS it is working and event variables are printed. But after a while (several hours of working) FS segfaults to this: kernel: [13834853.520624] freeswitch[31414]: segfault at 7f5b05178ca0 ip 00007f5b08100f32 sp 00007f5ae7d32d38 error 4 in libc-2.11.3.so[7f5b08085000+159000] If I comment , then FS runs ok without segfaults. My questions: 1) What is the better way to parse events: using start-up script which will hook to events and waits for them, or hook script to event in lua.conf and run the script only when event arise? 2) Why it segfaults on this script? Same code (function and getting variables) in other scripts, started from dialplan, does not segfault FS. 3) Was this a known bug and latest stable version will resolve the problem? I would not want to upgrade if the problem will remain. Thank you. -- Mimiko desu. From decipher.hk at gmail.com Sun Feb 7 21:18:27 2016 From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=) Date: Sun, 07 Feb 2016 18:18:27 +0000 Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter In-Reply-To: References: <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com> Message-ID: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com> February 5 2016 3:45 PM, "?talo Rossi" wrote: > Hi Rodrigo, Hi ?talo, > We don't keep these counters in memory. Ok. > If you want to figure out right now how many calls were abandoned or answered you need to parse the > cdrs from these calls or listen to the ESL events and keep a counter. > I see for agents we have no_answer_count and calls_answered > But, adding realtime counters to mod_callcenter shouldn't be difficult, a PR would be awesome ;) I'll try. What is the maximum line length columns?. In the [1] coding guidelines dont find 1: https://freeswitch.org/confluence/display/FREESWITCH/Coding+Guidelines -- Rodrigo Ram?rez Norambuena http://www.rodrigoramirez.com From shane.mitchell at fonedynamics.com.au Sun Feb 7 05:37:27 2016 From: shane.mitchell at fonedynamics.com.au (Shane Mitchell) Date: Sun, 7 Feb 2016 02:37:27 +0000 Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy Message-ID: Hi everyone, I'm after a bit of mod_managed help from those with experience. To explore how we can use mod_managed, I'm trying to simply get it up-and-running. To test, I'm trying to run Demo.csx (from source), however an ExecutionEngineException (see below) is always thrown whenever trying to load the csx file. I'm using latest stable Debian Jessie and FreeSWITCH (from packages). While I've been programming in C# since 1.0, I've never used mono, so I simply installed mono-complete from packages. I have also tested mono (and FreeSWITCH) successfully. I have researched related errors online with no luck or inspiration for a solution. Does anyone know why I would get this problem? Do I need to create any symlinks, install any packages, create directories (other than conf/mod/managed), etc to get mod_managed to work on a new install? Thanks all, much appreciated. [ALERT] switch_cpp.cpp:1356 Exception loading /usr/lib/freeswitch/mod/managed/Demo.csx: System.ExecutionEngineException: Failed to create shadow copy (ensure directory exists). Server stack trace: at (wrapper managed-to-native) System.AppDomain:LoadAssembly (System.AppDomain,string,System.Security.Policy.Evidence,bool) at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity, Boolean refonly) [0x00000] in :0 at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in :0 at (wrapper remoting-invoke-with-check) System.AppDomain:Load (string,System.Security.Policy.Evidence) at System.Reflection.Assembly.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in :0 at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, Boolean ignoreCase, BindingFlags bindingAttr, System.Reflection.Binder binder, System.Object[] args, System.Globalization.CultureInfo culture, System.Object[] activationAttributes, System.Security.Policy.Evidence securityInfo) [0x00000] in :0 at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 at System.AppDomain.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstance (string,string,object[]) at System.AppDomain.CreateInstanceAndUnwrap (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) at (wrapper xdomain-dispatch) System.AppDomain:CreateInstanceAndUnwrap (object,byte[]&,byte[]&,string,string) Exception rethrown at [0]: at (wrapper xdomain-invoke) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) at FreeSWITCH.Loader.loadFile (System.String fileName) [0x00000] in :0 From pskoul at gmail.com Sun Feb 7 23:00:26 2016 From: pskoul at gmail.com (Panagiotis Skoulikaritis) Date: Sun, 7 Feb 2016 22:00:26 +0200 Subject: [Freeswitch-users] extra header account code is not written to cdr if cancel is received a few ms after invite In-Reply-To: <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com> References: <56AE42E1.3080703@gmail.com> <3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com> <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com> Message-ID: <56B7A25A.8090801@gmail.com> I have tried both inline and export but I still have cdrs where the accountcode is not written. Any help would be greatly appreciated. Regards Panagiotis On 1/31/2016 11:58 PM, Oz Mortimer wrote: > Try export rather than set > >> On 31 Jan 2016, at 18:45, servtelar at gmail.com wrote: >> >> Shouldn't that be done as inline? >> >> Sent from my iPhone >> >>> On Jan 31, 2016, at 12:22 PM, Panagiotis Skoulikaritis wrote: >>> >>> Dear all >>> >>> I have an implementation FreeSWITCH as a sort of SBC, it is used to send >>> the calls to the terminating carriers and do topology hiding, nothing >>> fancy. Also I gather cdrs from the FreeSWITCH. >>> >>> In order to distinguish each customer on the FS cdrs I send an extra >>> header containing the accountcode. >>> >>> I have noticed that if the call is canceled immediately on the same sec, >>> the account code is not written on the cdr. >>> To be more precise the cancel is send a few milliseconds after it has >>> received the invite, and before the FreeSWITCH has sent the call to the >>> terminating carrier (I'm using Homer Sipcapture to capture all the >>> traces and I don't see an attempt being made at the terminating carrier) >>> also I don't see a b-leg cdr. >>> >>> FreeSWITCH is writing both a-leg and b-leg cdrs in csv format. >>> >>> The dialplan that I use is simple >>> >>> >>> >> expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> any idea how I can make sure that the account code will always be written ? >>> >>> >>> Best Regards >>> >>> Panagiotis >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From covici at ccs.covici.com Mon Feb 8 00:05:20 2016 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sun, 07 Feb 2016 16:05:20 -0500 Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy In-Reply-To: References: Message-ID: <16568.1454879120@ccs.covici.com> Do you have a managed directory under the mod directory (not sure where the package install puts that)? All my .dlls are in that directory. Shane Mitchell wrote: > Hi everyone, > > I'm after a bit of mod_managed help from those with experience. > > To explore how we can use mod_managed, I'm trying to simply get it up-and-running. To test, I'm trying to run Demo.csx (from source), however an ExecutionEngineException (see below) is always thrown whenever trying to load the csx file. > > I'm using latest stable Debian Jessie and FreeSWITCH (from packages). While I've been programming in C# since 1.0, I've never used mono, so I simply installed mono-complete from packages. I have also tested mono (and FreeSWITCH) successfully. I have researched related errors online with no luck or inspiration for a solution. > > Does anyone know why I would get this problem? Do I need to create any symlinks, install any packages, create directories (other than conf/mod/managed), etc to get mod_managed to work on a new install? > > Thanks all, much appreciated. > > > > [ALERT] switch_cpp.cpp:1356 Exception loading /usr/lib/freeswitch/mod/managed/Demo.csx: System.ExecutionEngineException: Failed to create shadow copy (ensure directory exists). > > Server stack trace: > at (wrapper managed-to-native) System.AppDomain:LoadAssembly (System.AppDomain,string,System.Security.Policy.Evidence,bool) > at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity, Boolean refonly) [0x00000] in :0 > at System.AppDomain.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in :0 > at (wrapper remoting-invoke-with-check) System.AppDomain:Load (string,System.Security.Policy.Evidence) > at System.Reflection.Assembly.Load (System.String assemblyString, System.Security.Policy.Evidence assemblySecurity) [0x00000] in :0 > at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, Boolean ignoreCase, BindingFlags bindingAttr, System.Reflection.Binder binder, System.Object[] args, System.Globalization.CultureInfo culture, System.Object[] activationAttributes, System.Security.Policy.Evidence securityInfo) [0x00000] in :0 > at System.Activator.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 > at System.AppDomain.CreateInstance (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 > at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstance (string,string,object[]) > at System.AppDomain.CreateInstanceAndUnwrap (System.String assemblyName, System.String typeName, System.Object[] activationAttributes) [0x00000] in :0 > at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) > at (wrapper xdomain-dispatch) System.AppDomain:CreateInstanceAndUnwrap (object,byte[]&,byte[]&,string,string) > > Exception rethrown at [0]: > > at (wrapper xdomain-invoke) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) > at (wrapper remoting-invoke-with-check) System.AppDomain:CreateInstanceAndUnwrap (string,string,object[]) > at FreeSWITCH.Loader.loadFile (System.String fileName) [0x00000] in :0 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From shane.mitchell at fonedynamics.com.au Mon Feb 8 02:00:35 2016 From: shane.mitchell at fonedynamics.com.au (Shane Mitchell) Date: Sun, 7 Feb 2016 23:00:35 +0000 Subject: [Freeswitch-users] mod_managed: Failed to create shadow copy In-Reply-To: <16568.1454879120@ccs.covici.com> References: <16568.1454879120@ccs.covici.com> Message-ID: >> Do you have a managed directory under the mod directory (not sure where the package install puts that)? All my .dlls are in that directory. I have created the managed directory as per the initial thread. After trial-and-error, I found the solution (simple really) to get mod_managed to work on a fresh install. It was simply a permissions problem. FS was trying to write to conf/mod/managed/ without sufficient permissions. So for those looking to try mod_managed, this is what I did to get it working: 1. Install mono. 2. Create conf/mod/managed with sufficient write permissions for FS. 3. Enable/add reference to mod_managed in modules.xml. 4. Add your csx/dll/etc files to conf/mod/managed. From stephen.thwaites at callstera.com Mon Feb 8 14:51:52 2016 From: stephen.thwaites at callstera.com (Stephen Thwaites) Date: Mon, 8 Feb 2016 12:51:52 +0100 Subject: [Freeswitch-users] BLF Subscriptions sometimes don't send an initial Notify Message-ID: Hello, I have setup presence and in most cases it is working as expected. i.e. Subscription is sent to FS, FS returns Accepted then immediately FS sends the notify to the phone, thereafter all the Notify events for ringing, pickup and hangup. Great. However in some cases a subscribe to an extension does subscribe,FS sends the accepted response but a Notify is not sent out at that point. However if I call the extension the Notify works perfectly. Any ideas of what could cause the initial Notify not to be sent after the Acceptance 202? Any help would be appreciated. Regards, Steve. p.s I sent this to the list a few days ago but it didn't seem to come through on the list. Some detail info below: FS is configured as Multi-Tennant # Multi-Tennant SIP Trace for Subscription FS Receives this from the phone: SUBSCRIBE sip:203 at xxx.mydomain.com :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport From: >;tag=946578510 To: >;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: :25060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.5.29 Expires: 480 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 FS Sends this back to the phone: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.24:25060 ;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x From: >;tag=946578510 To: >;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: :5060> Expires: 480 User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=480 Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/97a413fc/attachment.html From elvisnn at email.com Mon Feb 8 16:26:00 2016 From: elvisnn at email.com (Elvis) Date: Mon, 8 Feb 2016 06:26:00 -0700 (MST) Subject: [Freeswitch-users] FreeSWITCH abandons calls Message-ID: <1454937960255-7596200.post@n2.nabble.com> I can see form my logs that calls are abandoned with reason [CS_NEW] [WRONG_CALL_STATE]. Please see logs below: [WARNING] switch_core_state_machine.c:572 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e sofia/internal/61415158474 at 212.61.145.185 Abandoned [NOTICE] switch_core_state_machine.c:575 Hangup sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] [NOTICE] switch_core_session.c:1642 Session 8 (sofia/internal/61415158474 at 212.61.145.185) Ended [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY] Can someone please help out? Regards Elvis -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Feb 8 18:30:08 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Feb 2016 09:30:08 -0600 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: <1454937960255-7596200.post@n2.nabble.com> References: <1454937960255-7596200.post@n2.nabble.com> Message-ID: This usually means that a device has sent an invite, we replied with a challenge, but the device probably didn't receive our challenge. 'sofia global siptrace on' and watch it. On Mon, Feb 8, 2016 at 7:26 AM, Elvis wrote: > I can see form my logs that calls are abandoned with reason [CS_NEW] > [WRONG_CALL_STATE]. Please see logs below: > > [WARNING] switch_core_state_machine.c:572 > 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e > sofia/internal/61415158474 at 212.61.145.185 Abandoned > [NOTICE] switch_core_state_machine.c:575 Hangup > sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] > [NOTICE] switch_core_session.c:1642 Session 8 > (sofia/internal/61415158474 at 212.61.145.185) Ended > [NOTICE] switch_core_session.c:1646 Close Channel > sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY] > > > Can someone please help out? > > Regards > Elvis > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/ebef6258/attachment.html From krice at freeswitch.org Mon Feb 8 19:04:42 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 8 Feb 2016 10:04:42 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> <628620366.20160203092011@seznam.cz> <590d01d15e6a$22651810$672f4830$@freeswitch.org> <1047097771.103803.1454517591330.JavaMail.zimbra@voismart.it> Message-ID: This ticket was already resolved On Wed, Feb 3, 2016 at 10:39 AM, Davide Colombo wrote: > I reported this bug to jira: > > https://freeswitch.org/jira/browse/FS-8805 > > > > ----- Messaggio originale ----- > Da: "Ken Rice" > A: "freeswitch-users" > Inviato: Mercoled?, 3 febbraio 2016 11:03:29 > Oggetto: Re: [Freeswitch-users] Verto vs. SIP.js > > Re: [Freeswitch-users] Verto vs. SIP.js > > > Oh So where is the Jira on this it doesn?t work in firefox with the > debugging information? > > > > If you know of an issue like this you should report it to jira ( > https://freeswitch.org/jira ) so a dev can try to replicate and fix it? > > > > You cant expect bugs to get fixed if you aren?t reporting them properly? > > > > Go troll somewhere else?. > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev > Sent: Wednesday, February 3, 2016 2:20 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Verto vs. SIP.js > > > > > Dobr? den, > > The fact that Verto normally does not work on me Firefox does joy for me. > For example, I have all the stops on "Refresh Media Devices" (the picture > in the attachment) > My clients use different browsers and I can not force them to use > something specific. > With sip.js I have no such problems. It works perfectly, wherever there is > support webrtc without problems. > > > -- > S pozdravem, > Ing.Denis Jakovlev > mob.tel. 775-415-382 > > pond?l? 1. ?nora 2016, 17:22:24, napsal jste: > > With Verto you can also be more creative, see for example: > https://freeswitch.org/verto-not-just-for-call-signaling/ > It also provides an interesting feature: "attach". e.g. if a tab is closed > by mistake, FS can resume automatically the session upon reconnection (with > the same ID). > Also being a simple JSON-based protocol you can have people working on the > client side even with limited knowledge of SIP. > > On 1 February 2016 at 17:12, Ken Rice < krice at freeswitch.org > wrote: > Really? That?s why the whole protocol is OpenSource? not much vendor lock > in there? it can be used anywhere? > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Denis > Jakovlev > Sent: Monday, February 1, 2016 9:30 AM > To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > i think this Verto its too much VendorLock :) > > -- > S pozdravem, > Ing.Denis Jakovlev > mob.tel . 775-415-382 > > pond?l? 1. ?nora 2016, 15:42:48, napsal jste: > > Denis, what difficulties did you experience with Verto? > > Thanks. > > Bote > > > From: Denis Jakovlev > Sent: Monday, 01 February, 2016 09:26 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > I use sip.js. It turned out to be a lot easier that Verto. > > -- > S pozdravem, > Ing.Denis Jakovlev > mob.tel . 775-415-382 > > pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > > > Hi, > > > I want to setup a Click-2-Call-Button for our website. Is there any > > significant difference between Verto-Mod and libraries such as SIP.js? > > > I do not really understand the advantage of Verto (if there is any). > > > Thanks in advance, > > > Thorsten > > > _________________________________________________________________________ > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/93c30ae2/attachment-0001.html From DEdwards at vertical.com Mon Feb 8 19:36:50 2016 From: DEdwards at vertical.com (Dan Edwards) Date: Mon, 8 Feb 2016 16:36:50 +0000 Subject: [Freeswitch-users] WebSocket behind NGINX In-Reply-To: <56B667DA.4010505@gmail.com> References: <56AD0CE7.6000607@gmail.com> <56B667DA.4010505@gmail.com> Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997CE62@PHXEX2.vertical.com> Anton, I'm glad my input was useful. As for WSS vs WS, the fact you're using security bubbles up into the SIP messages themselves. I initially tried: Browser >> WSS >> Nginx >> WS >> FS FS does not like this because the protocol changes. You go from SIP/2.0/WSS to SIP/2.0/WS and FS won't allow that. Also, in some instances, you will get SIP URL changes. For example: sip:1234 at domain.com vs. sips:1234 at domain.com. The reason to go with WS to FS was to skip an encrypt/decrypt cycle on network traffic that never left the machine. I finally decided that trying to patch the SIP traffic was bound to fail at some point and we're only saving the encrypt/decrypt on the SIP traffic itself, so I went back to Browser >> WSS >> Nginx >>> WSS >> FS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton Sent: Saturday, February 06, 2016 4:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WebSocket behind NGINX Hi, Sorry for not answering for a long time. Dan, thank you, your recommendation really helped me. So in order to proxy websocket request you need: 1. Proxy websocket requests in this way WSS -> (NGINX) -> FS WSS or WS -> (NGINX) -> FS WS 2. Modify local-network-acl 3. Modify apply-candidate-acl if you would like to drop more rtp candidates PS: I highly recommend to watch this video about NAT issues and ACL configuration: https://www.youtube.com/watch?v=_WSx-T6TriI BR, Anton Voylenko On 01/30/2016 09:20 PM, Anton wrote: > Hello All, > > I have to proxy all websocket requests though a nginx server. Right > now I am using next configuration: > > map $http_upgrade $connection_upgrade { > default upgrade; > '' close; > } > > server { > listen 443; > server_name wss.somedomain.com.ua; > > ssl on; > ssl_certificate /etc/nginx/cert.pem; > ssl_certificate_key /etc/nginx/private.key; > > location / { > proxy_pass http://127.0.0.1:5066; > proxy_http_version 1.1; > proxy_set_header Upgrade $http_upgrade; > proxy_set_header Connection $connection_upgrade; > proxy_read_timeout 86400s; > } > > access_log /var/log/nginx/wss_access; > error_log /var/log/nginx/wss_error debug; } > > I dumped traffic from nginx and found out that "switching protocol" > phrase was successful but INVITE message from my browser in pending > state. > Maybe FreeSWITCH wants real IP not loopback? Who have faced with > similar problem? > > BR, > Anton _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Feb 8 22:18:42 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 8 Feb 2016 13:18:42 -0600 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: <628620366.20160203092011@seznam.cz> References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> <628620366.20160203092011@seznam.cz> Message-ID: On Wed, Feb 3, 2016 at 2:20 AM, Denis Jakovlev wrote: > Dobr? den, > > The fact that Verto normally does not work on me Firefox does joy for me. > For example, I have all the stops on "Refresh Media Devices" (the picture > in the attachment) > My clients use different browsers and I can not force them to use > something specific. > With sip.js I have no such problems. It works perfectly, wherever there is > support webrtc without problems. > > > You are comparing the Verto Communicator reference app with a sip.js library. Apples and Oranges. You found a bug in that app not the verto lib. The app developers have fixed the bug. The best advice is keep quiet here just like you do with reporting bugs and do not tout propaganda about one thing vs another. The answer to OP is that either one will work..... > > > > > > > > *-- S pozdravem, Ing.Denis Jakovlev mob.tel > . 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste: > * > > With Verto you can also be more creative, see for example: > https://freeswitch.org/verto-not-just-for-call-signaling/ > It also provides an interesting feature: "attach". e.g. if a tab is closed > by mistake, FS can resume automatically the session upon reconnection (with > the same ID). > Also being a simple JSON-based protocol you can have people working on the > client side even with limited knowledge of SIP. > > On 1 February 2016 at 17:12, Ken Rice wrote: > Really? That?s why the whole protocol is OpenSource? not much vendor lock > in there? it can be used anywhere? > > > > *From: *freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis > Jakovlev > *Sent:* Monday, February 1, 2016 9:30 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > i think this Verto its too much VendorLock :) > > > > > *-- S pozdravem, Ing.Denis Jakovlev *mob.tel > > > *. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: * > > Denis, what difficulties did you experience with Verto? > > Thanks. > > Bote > > > *From:* Denis Jakovlev > *Sent:* Monday, 01 February, 2016 09:26 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto vs. SIP.js > > Dobr? den, > > I use sip.js. It turned out to be a lot easier that Verto. > > > > > *-- S pozdravem, Ing.Denis Jakovlev *mob.tel > > > > > > > > > > > > > > > *. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I > want to setup a Click-2-Call-Button for our website. Is there any > > significant difference between Verto-Mod and libraries such as SIP.js? > I > do not really understand the advantage of Verto (if there is any). > Thanks > in advance, > Thorsten > > _________________________________________________________________________* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/3749f1ef/attachment.html From luis.daniel.lucio at gmail.com Mon Feb 8 17:16:00 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 8 Feb 2016 09:16:00 -0500 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: <1454937960255-7596200.post@n2.nabble.com> References: <1454937960255-7596200.post@n2.nabble.com> Message-ID: Turn on debug, you will get more detailed information. Maybe answer is there Le 8 f?vr. 2016 8:46 AM, "Elvis" a ?crit : > I can see form my logs that calls are abandoned with reason [CS_NEW] > [WRONG_CALL_STATE]. Please see logs below: > > [WARNING] switch_core_state_machine.c:572 > 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e > sofia/internal/61415158474 at 212.61.145.185 Abandoned > [NOTICE] switch_core_state_machine.c:575 Hangup > sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] > [NOTICE] switch_core_session.c:1642 Session 8 > (sofia/internal/61415158474 at 212.61.145.185) Ended > [NOTICE] switch_core_session.c:1646 Close Channel > sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY] > > > Can someone please help out? > > Regards > Elvis > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/59610cbe/attachment-0001.html From mbgatherer at gmail.com Tue Feb 9 00:21:19 2016 From: mbgatherer at gmail.com (Maciej Bylica) Date: Mon, 8 Feb 2016 22:21:19 +0100 Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP Message-ID: Hi All, I am looking for a class 5 platform (basic VAS) and softphone (IOS, Android) both supporting ZRTP protocol to achieve the highest voice security. C.5 and UA should be delivered from the same supplier (like sipwise for instance) Could anybody recommend me any solution here? Thanks in advanced -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/96e9884d/attachment.html From rtreleaven at bunnykick.ca Tue Feb 9 00:49:58 2016 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Mon, 8 Feb 2016 16:49:58 -0500 Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP In-Reply-To: References: Message-ID: This is the second mention of Class 5 in last couple of months. If you don't mind my asking why is Class 5 a requirement? Put another way is their a public specification that we should be referencing? Sincerely, Russell Treleaven On Mon, Feb 8, 2016 at 4:21 PM, Maciej Bylica wrote: > Hi All, > > I am looking for a class 5 platform (basic VAS) and softphone (IOS, > Android) both supporting ZRTP protocol to achieve the highest voice > security. > C.5 and UA should be delivered from the same supplier (like sipwise for > instance) > > Could anybody recommend me any solution here? > > Thanks in advanced > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/0a7b51f6/attachment.html From mbgatherer at gmail.com Tue Feb 9 01:15:48 2016 From: mbgatherer at gmail.com (Maciej Bylica) Date: Mon, 8 Feb 2016 23:15:48 +0100 Subject: [Freeswitch-users] Class 5 and softphone app supporting ZRTP In-Reply-To: References: Message-ID: Hi Thanks for prompt answer. By saying class5, i was referring to platform capable of delivering VoIP features to the end subscribers. Of course it could be FS+OS/Kamailio+DB+scripting+web As for the beginning i need basic features like Voicemail, IVR, Call Hold, CallWaiting, but wouldn't like to limit myself in the future. The app needs to be auto-provisioned once downloaded and installed. Push notification, in-app purchases and ZRTP is a must here. Thanks Maciej. 2016-02-08 22:49 GMT+01:00 Russell Treleaven : > This is the second mention of Class 5 in last couple of months. > If you don't mind my asking why is Class 5 a requirement? > Put another way is their a public specification that we should be > referencing? > > Sincerely, > > Russell Treleaven > > > On Mon, Feb 8, 2016 at 4:21 PM, Maciej Bylica > wrote: > >> Hi All, >> >> I am looking for a class 5 platform (basic VAS) and softphone (IOS, >> Android) both supporting ZRTP protocol to achieve the highest voice >> security. >> C.5 and UA should be delivered from the same supplier (like sipwise for >> instance) >> >> Could anybody recommend me any solution here? >> >> Thanks in advanced >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/707607c5/attachment.html From netcentrica at gmail.com Tue Feb 9 01:40:42 2016 From: netcentrica at gmail.com (Mateusz Bartczak) Date: Mon, 8 Feb 2016 23:40:42 +0100 Subject: [Freeswitch-users] Enterprise/sequential originate different stop codes per gateway Message-ID: Hi all I'm looking for a proper way to achieve functionality of different stop codes per every called gateway Example call flow: call provider 1, stop on code 404 if provider 1 fails call provider 2, stop on code 404,486 if provider 2 fails call provider 3, stop on code 480 and so on I'm aware of variable continue_on_fail but I had no luck in specifying this per gateway, it seems to work only for whole originate string, not individual gateways Could you share some examples how to do this? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/3f47a15d/attachment.html From s.safarov at gmail.com Tue Feb 9 07:41:01 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 9 Feb 2016 07:41:01 +0300 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: References: <1454937960255-7596200.post@n2.nabble.com> Message-ID: Probably you server has public IP address. In this case hackers searching open servers or account with out password registration. FS-7125 PR-159 On Mon, Feb 8, 2016 at 5:16 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Turn on debug, you will get more detailed information. Maybe answer is > there > Le 8 f?vr. 2016 8:46 AM, "Elvis" a ?crit : > >> I can see form my logs that calls are abandoned with reason [CS_NEW] >> [WRONG_CALL_STATE]. Please see logs below: >> >> [WARNING] switch_core_state_machine.c:572 >> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e >> sofia/internal/61415158474 at 212.61.145.185 Abandoned >> [NOTICE] switch_core_state_machine.c:575 Hangup >> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] >> [NOTICE] switch_core_session.c:1642 Session 8 >> (sofia/internal/61415158474 at 212.61.145.185) Ended >> [NOTICE] switch_core_session.c:1646 Close Channel >> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY] >> >> >> Can someone please help out? >> >> Regards >> Elvis >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/84415ddf/attachment-0001.html From idokan at gmail.com Tue Feb 9 08:42:22 2016 From: idokan at gmail.com (ik) Date: Tue, 9 Feb 2016 07:42:22 +0200 Subject: [Freeswitch-users] Capturing leg B hangup from bridge Message-ID: Hello, I'm trying to use hangup api hook for leg b, while leg a is still connected and transferred to do other tasks. The hook only triggers when leg a disconnected, not when leg b. I found this question on Google, but no answer that works. So how can I execute API (curl in this case) when leg b is disconnected? Thank you, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/f58ef240/attachment.html From idokan at gmail.com Tue Feb 9 11:09:54 2016 From: idokan at gmail.com (ik) Date: Tue, 9 Feb 2016 10:09:54 +0200 Subject: [Freeswitch-users] Capturing leg B hangup from bridge In-Reply-To: References: Message-ID: Answering myself : export and nolocal to do it. On Feb 9, 2016 7:42 AM, "ik" wrote: > Hello, > > I'm trying to use hangup api hook for leg b, while leg a is still > connected and transferred to do other tasks. > > The hook only triggers when leg a disconnected, not when leg b. > > I found this question on Google, but no answer that works. > > So how can I execute API (curl in this case) when leg b is disconnected? > > Thank you, > > Ido > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/0d8dcf7f/attachment.html From achinthau at gmail.com Tue Feb 9 15:39:01 2016 From: achinthau at gmail.com (Achintha) Date: Tue, 9 Feb 2016 18:09:01 +0530 Subject: [Freeswitch-users] video conference not working Message-ID: hi all, We are checking SIP video conferencing on FreeSwitch through mod_conference but the video is not getting enabled. Video works fine on a normal user to user call and codecs are matching. But for the conference room we create, the softphone displays an error message saying ?No matching video codecs found?. We are using FreeSwitch version 1.6 kindly advice me to solve this problem Thanking you Achintha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/5c2b244d/attachment.html From gmaruzz at gmail.com Tue Feb 9 16:21:33 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 9 Feb 2016 14:21:33 +0100 Subject: [Freeswitch-users] video conference not working In-Reply-To: References: Message-ID: you must pastebin the dialplan, the conference settings, the complete debug from fs_cli how we can know what is happening? On Tue, Feb 9, 2016 at 1:39 PM, Achintha wrote: > hi all, > > > We are checking SIP video conferencing on FreeSwitch through > mod_conference but the video is not getting enabled. Video works fine on a > normal user to user call and codecs are matching. But for the conference > room we create, the softphone displays an error message saying ?No matching > video codecs found?. We are using FreeSwitch version 1.6 > kindly advice me to solve this problem > > Thanking you > Achintha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/2adaed39/attachment.html From elvisnn at email.com Tue Feb 9 16:37:02 2016 From: elvisnn at email.com (Elvis N. Ngah) Date: Tue, 9 Feb 2016 14:37:02 +0100 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: References: <1454937960255-7596200.post@n2.nabble.com>, Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/30ab3a9e/attachment.html From brian at freeswitch.org Tue Feb 9 16:42:48 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Feb 2016 07:42:48 -0600 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: References: <1454937960255-7596200.post@n2.nabble.com> Message-ID: You've got a nat problem, disabling this won't probably fix your issue! What devices are involved and network topology? /b On Tuesday, February 9, 2016, Elvis N. Ngah wrote: > Thank you Brian! > > I did so and saw exactly what you told me. The INVITE is being challenged > with a 407 - Proxy Authentication Required. I would like to disable this > feature. Can you please let me know where to do that? > > Regards > Elvis > > *Sent:* Monday, February 08, 2016 at 4:30 PM > *From:* "Brian West" > > *To:* "FreeSWITCH Users Help" > > *Subject:* Re: [Freeswitch-users] FreeSWITCH abandons calls > This usually means that a device has sent an invite, we replied with a > challenge, but the device probably didn't receive our challenge. 'sofia > global siptrace on' and watch it. > > > > On Mon, Feb 8, 2016 at 7:26 AM, Elvis wrote: >> >> I can see form my logs that calls are abandoned with reason [CS_NEW] >> [WRONG_CALL_STATE]. Please see logs below: >> >> [WARNING] switch_core_state_machine.c:572 >> 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e >> sofia/internal/61415158474 at 212.61.145.185 Abandoned >> [NOTICE] switch_core_state_machine.c:575 Hangup >> sofia/internal/61415158474 at 212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] >> [NOTICE] switch_core_session.c:1642 Session 8 >> (sofia/internal/61415158474 at 212.61.145.185) Ended >> [NOTICE] switch_core_session.c:1646 Close Channel >> sofia/internal/61415158474 at 212.61.145.185 [CS_DESTROY] >> >> >> Can someone please help out? >> >> Regards >> Elvis >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7596200.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > > http://www.freeswitchsolutions.com Official FreeSWITCH Sites > http://www.freeswitch.org http://confluence.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/8c8efda3/attachment-0001.html From bilaln018 at gmail.com Tue Feb 9 16:51:51 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 9 Feb 2016 18:51:51 +0500 Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs] Message-ID: Hi all, I just want to enable the wss on freeswitch, so i have configuration in place, wss listening on port 7443, wss.pem is added in the certs directory, but i am unable to connect using https://tryit.jssip.net/,(i can get registered using wss://tryit.jssip.net:10443 but not through wss://MYSERVERIP:7443) So my question is how can i check the debug logs that whats actually happening on the switch, i can see the hits coming on server using tcpdump. Any help will be highly appreciated. Thanks Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/626ae5f6/attachment.html From brian at freeswitch.org Tue Feb 9 17:14:40 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Feb 2016 08:14:40 -0600 Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs] In-Reply-To: References: Message-ID: Did you setup your certificates? if you try to visit https://MYSERVERIP:7443 what do you get? On Tue, Feb 9, 2016 at 7:51 AM, Bilal Abbasi wrote: > Hi all, > I just want to enable the wss on freeswitch, so i have configuration in > place, wss listening on port 7443, wss.pem is added in the certs directory, > but i am unable to connect using https://tryit.jssip.net/,(i can get > registered using wss://tryit.jssip.net:10443 but not through > wss://MYSERVERIP:7443) > So my question is how can i check the debug logs that whats actually > happening on the switch, i can see the hits coming on server using tcpdump. > Any help will be highly appreciated. > > Thanks > Abbasi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/ab44e977/attachment.html From DEdwards at vertical.com Tue Feb 9 17:18:01 2016 From: DEdwards at vertical.com (Dan Edwards) Date: Tue, 9 Feb 2016 14:18:01 +0000 Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs] In-Reply-To: References: Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com> If the certificates in wss.pem are not created for MYSERVERIP, this will fail. The easiest way to check is to enter the URL in your browsers? address bar, substituting https for wss (ie. https://MYSERVERIP:7443). If your browser throws up a security issue, the SSL certificates in wss.pem do not match your host. If that?s the case, you?ll need to purchase your own SSL certificates (if you want to make this publically available) or create self-signed certs for testing/debugging. See https://freeswitch.org/confluence/display/FREESWITCH/WebRTC for more info. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bilal Abbasi Sent: Tuesday, February 09, 2016 8:52 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs] Hi all, I just want to enable the wss on freeswitch, so i have configuration in place, wss listening on port 7443, wss.pem is added in the certs directory, but i am unable to connect using https://tryit.jssip.net/,(i can get registered using wss://tryit.jssip.net:10443 but not through wss://MYSERVERIP:7443) So my question is how can i check the debug logs that whats actually happening on the switch, i can see the hits coming on server using tcpdump. Any help will be highly appreciated. Thanks Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/0b51729e/attachment.html From astpp at inextrix.com Mon Feb 8 12:44:57 2016 From: astpp at inextrix.com (ASTPP Opensource VOIP Billing) Date: Mon, 8 Feb 2016 15:14:57 +0530 Subject: [Freeswitch-users] ASTPP Team launched Mobile SIP Dialer Message-ID: Hi Everyone, ASTPP Team glad to announce first ever Mobile SIP Dialer which is integrated with ASTPP. *ASTPP Dialer* is a brand new Mobile SIP Dialer launched by ASTPP Team. Its a complete SIP Softphone which allows you to register your SIP account on any SIP server, specially ASTPP server. *Features:* - Easy to use & User friendly graphical interface - Call Transfer, Call Hold - Integration with Mobile Phonebook - Fast registration and call connectivity - Works with any SIP server - Displays real time Account balance (ASTPP only) - Displays Call history - Connect through 3G, 4G, GPRS and Wi-Fi - G729, GSM, iLBC, Speex, G711, G722, AMR Codec Support *Download:* ASTPP Dialer is available freely in *Google play store*: https://play.google.com/store/apps/details?id=com.inextrix.astppdialer Please do not forget to share your review in play store. *Want your own branded ASTPP Dialer?* We do also offer complete integration with your ASTPP server, and *rebrand / customize it with your company name/logo* in both *Android and iOS*. To get more details on it please *Click Here *and send your requirements on sales at inextrix.com -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160208/d5e9c3c7/attachment-0001.html From yadenis at seznam.cz Tue Feb 9 10:47:26 2016 From: yadenis at seznam.cz (Denis Jakovlev) Date: Tue, 9 Feb 2016 08:47:26 +0100 Subject: [Freeswitch-users] Verto vs. SIP.js In-Reply-To: References: <56AF67C9.2010700@level5.de> <95599933.20160201152539@seznam.cz> <003a01d15cfe$d26c3350$774499f0$@botecomm.com> <1105687706.20160201163013@seznam.cz> <505001d15d0b$623021c0$26906540$@freeswitch.org> <628620366.20160203092011@seznam.cz> Message-ID: <1481152288.20160209084726@seznam.cz> Dobr? den, I'm sorry, but in order to try, as it generally works I try first reference app. If this does not work where I want, I just go on. No propaganda. For a button to call a site I stopped at sip.js. It's all. If I can not say this when asked, I apologize. More I will not do. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 8. ?nora 2016, 20:18:42, napsal jste: On Wed, Feb 3, 2016 at 2:20 AM, Denis Jakovlev wrote: Dobr? den, The fact that Verto normally does not work on me Firefox does joy for me. For example, I have all the stops on "Refresh Media Devices" (the picture in the attachment) My clients use different browsers and I can not force them to use something specific. With sip.js I have no such problems. It works perfectly, wherever there is support webrtc without problems. You are comparing the Verto Communicator reference app with a sip.js library. Apples and Oranges. You found a bug in that app not the verto lib. The app developers have fixed the bug. The best advice is keep quiet here just like you do with reporting bugs and do not tout propaganda about one thing vs another. The answer to OP is that either one will work..... -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 17:22:24, napsal jste: With Verto you can also be more creative, see for example: https://freeswitch.org/verto-not-just-for-call-signaling/ It also provides an interesting feature: "attach". e.g. if a tab is closed by mistake, FS can resume automatically the session upon reconnection (with the same ID). Also being a simple JSON-based protocol you can have people working on the client side even with limited knowledge of SIP. On 1 February 2016 at 17:12, Ken Rice wrote: Really? That?s why the whole protocol is OpenSource? not much vendor lock in there? it can be used anywhere? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Jakovlev Sent: Monday, February 1, 2016 9:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, i think this Verto its too much VendorLock :) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:42:48, napsal jste: Denis, what difficulties did you experience with Verto? Thanks. Bote From: Denis Jakovlev Sent: Monday, 01 February, 2016 09:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto vs. SIP.js Dobr? den, I use sip.js. It turned out to be a lot easier that Verto. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 pond?l? 1. ?nora 2016, 15:12:25, napsal jste: > Hi, > I want to setup a Click-2-Call-Button for our website. Is there any > significant difference between Verto-Mod and libraries such as SIP.js? > I do not really understand the advantage of Verto (if there is any). > Thanks in advance, > Thorsten > _________________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/1c6588c5/attachment-0001.html From elvisnn at email.com Tue Feb 9 17:47:22 2016 From: elvisnn at email.com (Elvis N. Ngah) Date: Tue, 9 Feb 2016 15:47:22 +0100 Subject: [Freeswitch-users] FreeSWITCH abandons calls In-Reply-To: References: <1454937960255-7596200.post@n2.nabble.com> , Message-ID: An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/bfbd00e0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: SIP Signaling.pdf Type: application/pdf Size: 81008 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/bfbd00e0/attachment-0001.pdf From royj at yandex.ru Tue Feb 9 18:10:29 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Tue, 09 Feb 2016 18:10:29 +0300 Subject: [Freeswitch-users] Outbound async ESL, linger, pickup Message-ID: <848891455030629@web25h.yandex.ru> Hi, all Is anybody used pickup ' https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+pickup ' in outbound async ESL, linger? When trying pickup for example after this: sendmsg execute-app-name: bridge execute-app-arg: sofia/gateway/gateway_name/number,pickup/100 call-command: execute with: sendmsg execute-app-name: pickup execute-app-arg: 100 call-command: execute I got +OK and immediately CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP_COMPLETE for channel that calls pickup. It is not a script issues hangup command. But if that pickup to call from dialplan like and bridge from ESL then all works as expected. Can somebody point what might be wrong here? Or clarify the picture. From gmaruzz at gmail.com Tue Feb 9 20:25:53 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 9 Feb 2016 18:25:53 +0100 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! Message-ID: Hey fellow FreeSWITCHers! How much we all made in 2015 thanks to FreeSWITCH? How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! On top right in http://www.freeswitch.org page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team. Core devs are having the annual development meeting days. If not now, when? Because! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/3cc9beaa/attachment.html From blake at cogents.io Tue Feb 9 21:26:45 2016 From: blake at cogents.io (Blake Priddy) Date: Tue, 9 Feb 2016 12:26:45 -0600 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: References: Message-ID: C'mon y'all!!! https://www.gofundme.com/freeswitch On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" wrote: > Hey fellow FreeSWITCHers! > > How much we all made in 2015 thanks to FreeSWITCH? > > How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, > etc) answering our difficult questions? > > Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! > > On top right in http://www.freeswitch.org page there is a "Donate" Paypal > button that will directly translate in gluttony for the Core FreeSWITCH > Team. > > Core devs are having the annual development meeting days. > > If not now, when? > > Because! > > -giovanni > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/4fbfdb04/attachment.html From thomas at bettervoice.com Wed Feb 10 05:23:17 2016 From: thomas at bettervoice.com (Thomas Quintana) Date: Tue, 9 Feb 2016 21:23:17 -0500 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: References: Message-ID: +1 Thomas Quintana Chief Technology Officer Phone: 1 (512) 677-6200 Website: http://www.bettervoice.com On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy wrote: > C'mon y'all!!! > > https://www.gofundme.com/freeswitch > On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" wrote: > >> Hey fellow FreeSWITCHers! >> >> How much we all made in 2015 thanks to FreeSWITCH? >> >> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, >> etc) answering our difficult questions? >> >> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us >> ! >> >> On top right in http://www.freeswitch.org page there is a "Donate" >> Paypal button that will directly translate in gluttony for the Core >> FreeSWITCH Team. >> >> Core devs are having the annual development meeting days. >> >> If not now, when? >> >> Because! >> >> -giovanni >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/556313e3/attachment.html From bilaln018 at gmail.com Wed Feb 10 08:03:17 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 10 Feb 2016 10:03:17 +0500 Subject: [Freeswitch-users] [WSS Unable to connect][No Debug logs] In-Reply-To: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997D3D1@PHXEX2.vertical.com> Message-ID: Hi Brain,Edwards, Thanks a lot for the comments, yes the issue was with wss.pem file, i have purchased the domain and certificate for my server, and it started working, Regards Abbasi On Tue, Feb 9, 2016 at 7:18 PM, Dan Edwards wrote: > If the certificates in wss.pem are not created for MYSERVERIP, this will > fail. > > > > The easiest way to check is to enter the URL in your browsers? address > bar, substituting https for wss (ie. https://MYSERVERIP:7443). If your > browser throws up a security issue, the SSL certificates in wss.pem do not > match your host. If that?s the case, you?ll need to purchase your own SSL > certificates (if you want to make this publically available) or create > self-signed certs for testing/debugging. > > > > See https://freeswitch.org/confluence/display/FREESWITCH/WebRTC for more > info. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bilal Abbasi > *Sent:* Tuesday, February 09, 2016 8:52 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] [WSS Unable to connect][No Debug logs] > > > > Hi all, > > I just want to enable the wss on freeswitch, so i have configuration in > place, wss listening on port 7443, wss.pem is added in the certs directory, > but i am unable to connect using https://tryit.jssip.net/,(i can get > registered using wss://tryit.jssip.net:10443 but not through > wss://MYSERVERIP:7443) > > So my question is how can i check the debug logs that whats actually > happening on the switch, i can see the hits coming on server using tcpdump. > > Any help will be highly appreciated. > > > > Thanks > > Abbasi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/c4e74ee3/attachment-0001.html From jprangi at didforsale.com Wed Feb 10 09:55:39 2016 From: jprangi at didforsale.com (Jai Rangi) Date: Tue, 9 Feb 2016 22:55:39 -0800 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: References: Message-ID: +2 *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana wrote: > +1 > > Thomas Quintana > Chief Technology Officer > Phone: 1 (512) 677-6200 > Website: http://www.bettervoice.com > > > > On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy wrote: > >> C'mon y'all!!! >> >> https://www.gofundme.com/freeswitch >> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" wrote: >> >>> Hey fellow FreeSWITCHers! >>> >>> How much we all made in 2015 thanks to FreeSWITCH? >>> >>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, >>> William, etc) answering our difficult questions? >>> >>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on >>> us ! >>> >>> On top right in http://www.freeswitch.org page there is a "Donate" >>> Paypal button that will directly translate in gluttony for the Core >>> FreeSWITCH Team. >>> >>> Core devs are having the annual development meeting days. >>> >>> If not now, when? >>> >>> Because! >>> >>> -giovanni >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160209/883e1103/attachment.html From blake at cogents.io Wed Feb 10 11:07:10 2016 From: blake at cogents.io (Blake Priddy) Date: Wed, 10 Feb 2016 02:07:10 -0600 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: References: Message-ID: +3 https://www.gofundme.com/freeswitch On Feb 10, 2016 1:00 AM, "Jai Rangi" wrote: > +2 > > *Jai Rangi* > Cebod Technologies LLC dba DIDforSale/Cebod Telecom > O 949-471-0102 | C 949-419-7634 | F > 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | > www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | > > > > > > > On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana > wrote: > >> +1 >> >> Thomas Quintana >> Chief Technology Officer >> Phone: 1 (512) 677-6200 >> Website: http://www.bettervoice.com >> >> >> >> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy wrote: >> >>> C'mon y'all!!! >>> >>> https://www.gofundme.com/freeswitch >>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" >>> wrote: >>> >>>> Hey fellow FreeSWITCHers! >>>> >>>> How much we all made in 2015 thanks to FreeSWITCH? >>>> >>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, >>>> William, etc) answering our difficult questions? >>>> >>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on >>>> us ! >>>> >>>> On top right in http://www.freeswitch.org page there is a "Donate" >>>> Paypal button that will directly translate in gluttony for the Core >>>> FreeSWITCH Team. >>>> >>>> Core devs are having the annual development meeting days. >>>> >>>> If not now, when? >>>> >>>> Because! >>>> >>>> -giovanni >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/27e05201/attachment-0001.html From sdame at 207me.com Wed Feb 10 16:20:17 2016 From: sdame at 207me.com (Stephen Dame) Date: Wed, 10 Feb 2016 08:20:17 -0500 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: References: Message-ID: <006501d16405$e4e71aa0$aeb54fe0$@207me.com> +4 Regards, Stephen HostBBB ? Online Learning Solutions http://www.hostbbb.com 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy Sent: Wednesday, February 10, 2016 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! +3 https://www.gofundme.com/freeswitch On Feb 10, 2016 1:00 AM, "Jai Rangi" > wrote: +2 Jai Rangi Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana > wrote: +1 Thomas Quintana Chief Technology Officer Phone: 1 (512) 677-6200 Website: http://www.bettervoice.com On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy > wrote: C'mon y'all!!! https://www.gofundme.com/freeswitch On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" > wrote: Hey fellow FreeSWITCHers! How much we all made in 2015 thanks to FreeSWITCH? How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! On top right in http://www.freeswitch.org page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team. Core devs are having the annual development meeting days. If not now, when? Because! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/7f0a5068/attachment.html From jjserranor at gmail.com Wed Feb 10 13:31:56 2016 From: jjserranor at gmail.com (Jose Serrano) Date: Wed, 10 Feb 2016 11:31:56 +0100 Subject: [Freeswitch-users] P-asserted-identity Message-ID: Hello. My freeswitch by default replace the P-asserted-identity by Remote-Party-ID when routing calls to the gateway. I want to send the P-asserted-identity and not the Remote-Party-ID and for that I tried the following: I have configured in the outbound gateway definition the following parameters: or but te behavior is the same. The only thing that works is to set up the "sip_cid_type" variable in the dialplan like this: Anyone knows how I can send the P-Asserted-identity without having to modify all my dial plan adding the sip_cid_type? Thanks in avanced -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/a18a9d12/attachment.html From emamirazavi at gmail.com Wed Feb 10 17:18:52 2016 From: emamirazavi at gmail.com (S.Mohammad Emami Razavi) Date: Wed, 10 Feb 2016 17:48:52 +0330 Subject: [Freeswitch-users] mod_verto and a1-hash Message-ID: Hello, I'm generating hashed password in the domain and with the username in md5. But after setting it as a1-hash param in user directory, verto user can not authenticate correctly and an error is returned from mod_verto. After all authenticating with plain username and password and without a1-hash set in user directory param, is excellent in mod_verto. Can anybody help?! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/796b771d/attachment-0001.html From v.kovalyshyn at gmail.com Wed Feb 10 17:23:54 2016 From: v.kovalyshyn at gmail.com (Vitaly Kovalyshyn) Date: Wed, 10 Feb 2016 16:23:54 +0200 Subject: [Freeswitch-users] mod_verto and a1-hash In-Reply-To: References: Message-ID: https://freeswitch.org/jira/browse/FS-6982 Best regards, Vitaly Kovalyshyn > On 10 ???. 2016 ?., at 16:18, S.Mohammad Emami Razavi wrote: > > Hello, I'm generating hashed password in the domain and with the username in md5. But after setting it as a1-hash param in user directory, verto user can not authenticate correctly and an error is returned from mod_verto. After all authenticating with plain username and password and without a1-hash set in user directory param, is excellent in mod_verto. > Can anybody help?! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/ac94ce9c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3701 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/ac94ce9c/attachment.bin From royj at yandex.ru Wed Feb 10 17:33:43 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Wed, 10 Feb 2016 17:33:43 +0300 Subject: [Freeswitch-users] P-asserted-identity In-Reply-To: References: Message-ID: <2566771455114823@web13h.yandex.ru> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/64d868ba/attachment.html From mike at jerris.com Wed Feb 10 18:54:29 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Feb 2016 09:54:29 -0600 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: <006501d16405$e4e71aa0$aeb54fe0$@207me.com> References: <006501d16405$e4e71aa0$aeb54fe0$@207me.com> Message-ID: <645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com> while (x++) { printf("Thanks Everyone!!!\n"); if (x == MAX_UINT64T) { printf("Whoa!!!\n"); } } > On Feb 10, 2016, at 7:20 AM, Stephen Dame wrote: > > +4 > > Regards, > Stephen > > HostBBB ? Online Learning Solutions http://www.hostbbb.com > 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy > Sent: Wednesday, February 10, 2016 3:07 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! > > +3 > > https://www.gofundme.com/freeswitch > On Feb 10, 2016 1:00 AM, "Jai Rangi" > wrote: >> +2 >> >> Jai Rangi >> Cebod Technologies LLC dba DIDforSale/Cebod Telecom >> O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | >> >> >> >> >> >> >> On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana > wrote: >>> +1 >>> >>> Thomas Quintana >>> Chief Technology Officer >>> Phone: 1 (512) 677-6200 >>> Website: http://www.bettervoice.com >>> >>> >>> >>> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy > wrote: >>>> C'mon y'all!!! >>>> >>>> https://www.gofundme.com/freeswitch >>>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" > wrote: >>>>> Hey fellow FreeSWITCHers! >>>>> >>>>> How much we all made in 2015 thanks to FreeSWITCH? >>>>> >>>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? >>>>> >>>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! >>>>> >>>>> On top right in http://www.freeswitch.org page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team. >>>>> >>>>> Core devs are having the annual development meeting days. >>>>> >>>>> If not now, when? >>>>> >>>>> Because! >>>>> >>>>> -giovanni >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/f8e987ac/attachment-0001.html From mike at jerris.com Wed Feb 10 18:56:47 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Feb 2016 09:56:47 -0600 Subject: [Freeswitch-users] P-asserted-identity In-Reply-To: References: Message-ID: I think the other response answers your question, however I wanted to clarify one thing. FreeSWITCH is a B2BUA. It is not "replacing" the PAID with RPID, its a new call that is sent with its own configuration. Mike > On Feb 10, 2016, at 4:31 AM, Jose Serrano wrote: > > Hello. > > My freeswitch by default replace the P-asserted-identity by Remote-Party-ID when routing calls to the gateway. > I want to send the P-asserted-identity and not the Remote-Party-ID and for that I tried the following: > > I have configured in the outbound gateway definition the following parameters: > or > > but te behavior is the same. > > The only thing that works is to set up the "sip_cid_type" variable in the dialplan like this: > > > Anyone knows how I can send the P-Asserted-identity without having to modify all my dial plan adding the sip_cid_type? > > Thanks in avanced > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nneul at mst.edu Wed Feb 10 19:15:28 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 10 Feb 2016 10:15:28 -0600 Subject: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! In-Reply-To: <645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com> References: <006501d16405$e4e71aa0$aeb54fe0$@207me.com> <645A5847-61E2-42CE-94B4-EDF5AD7C6FD2@jerris.com> Message-ID: <56BB6220.3040605@mst.edu> Better be careful there, that next loop takes all the money back. :) -- Nathan On 02/10/2016 09:54 AM, Michael Jerris wrote: > while (x++) { > printf("Thanks Everyone!!!\n"); > if (x == MAX_UINT64T) { > printf("Whoa!!!\n"); > } > } > >> On Feb 10, 2016, at 7:20 AM, Stephen Dame > wrote: >> >> +4 >> Regards, >> Stephen >> HostBBB ? Online Learning Solutions http://www.hostbbb.com >> 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]*On Behalf Of*Blake Priddy >> *Sent:*Wednesday, February 10, 2016 3:07 AM >> *To:*FreeSWITCH Users Help > >> *Subject:*Re: [Freeswitch-users] Hit that wallet now ! FreeSWITCH core lunch and dinner ! >> >> +3 >> >> https://www.gofundme.com/freeswitch >> >> On Feb 10, 2016 1:00 AM, "Jai Rangi" > wrote: >>> +2 >>> >>> *Jai Rangi* >>> Cebod Technologies LLC dba DIDforSale/Cebod Telecom >>> O 949-471-0102 |C 949-419-7634 |F 949-269-0449 / >>> 949-232-1410|jprangi at didforsale.com www.cebod.com >>> |www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA >>> 92626 | >>> >>> >>> >>> >>> >>> On Tue, Feb 9, 2016 at 6:23 PM, Thomas Quintana > wrote: >>>> +1 >>>> >>>> Thomas Quintana >>>> Chief Technology Officer >>>> Phone:1 (512) 677-6200 >>>> Website:http://www.bettervoice.com >>>> On Tue, Feb 9, 2016 at 1:26 PM, Blake Priddy > wrote: >>>>> >>>>> C'mon y'all!!! >>>>> >>>>> https://www.gofundme.com/freeswitch >>>>> >>>>> On Feb 9, 2016 12:22 PM, "Giovanni Maruzzelli" > wrote: >>>>>> >>>>>> Hey fellow FreeSWITCHers! >>>>>> >>>>>> How much we all made in 2015 thanks to FreeSWITCH? >>>>>> >>>>>> How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? >>>>>> >>>>>> Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! >>>>>> >>>>>> On top right inhttp://www.freeswitch.org page there is a "Donate" Paypal button that >>>>>> will directly translate in gluttony for the Core FreeSWITCH Team. >>>>>> >>>>>> Core devs are having the annual development meeting days. >>>>>> If not now, when? >>>>>> >>>>>> Because! >>>>>> >>>>>> -giovanni >>>>>> >>>>>> -- >>>>>> Sincerely, >>>>>> >>>>>> Giovanni Maruzzelli >>>>>> Cell :+39-347-2665618 >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From thomas.granvej6 at gmail.com Wed Feb 10 19:48:22 2016 From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=) Date: Wed, 10 Feb 2016 17:48:22 +0100 Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8 Message-ID: Hi all, Using 1.6.2 this works from fs_cli and dialplan: originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. Has something changed between 1.6.2 and 1.6.6 that may be the cause of this, and if so, what can I do to fix it? :o) Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/4b1f7053/attachment.html From krice at freeswitch.org Wed Feb 10 20:07:30 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 10 Feb 2016 11:07:30 -0600 Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8 In-Reply-To: References: Message-ID: I doubt that was ever an intended feature... why not just send the A leg out and park the bleg just reverse your notation there On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke wrote: > Hi all, > > Using 1.6.2 this works from fs_cli and dialplan: > > originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) > > Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. > > Has something changed between 1.6.2 and 1.6.6 that may be the cause of > this, and if so, what can I do to fix it? > > :o) > Thomas > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/a79e1ab9/attachment.html From thomas.granvej6 at gmail.com Wed Feb 10 20:16:19 2016 From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=) Date: Wed, 10 Feb 2016 18:16:19 +0100 Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8 In-Reply-To: References: Message-ID: Hi Ken, It doesn't work from our dialplan either: That works swimmingly with 1.6.2, but fails with BEARERCAPABILITY_NOTAVAIL with 1.6.6. 2016-02-10 18:07 GMT+01:00 Ken Rice : > I doubt that was ever an intended feature... why not just send the A leg > out and park the bleg just reverse your notation there > > On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke > wrote: > >> Hi all, >> >> Using 1.6.2 this works from fs_cli and dialplan: >> >> originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) >> >> Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/c534b44b/attachment-0001.html From brian at freeswitch.org Wed Feb 10 20:18:42 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Feb 2016 11:18:42 -0600 Subject: [Freeswitch-users] BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8 In-Reply-To: References: Message-ID: Look at the sip signaling On Wednesday, February 10, 2016, Thomas L?cke wrote: > Hi Ken, > > It doesn't work from our dialplan either: > > > expression="^external_transfer_(\d+)$"> > > > data="[fifo_music=default]sofia/gateway/${default_trunk}/$1"/> > > > > > That works swimmingly with 1.6.2, but fails with BEARERCAPABILITY_NOTAVAIL > with 1.6.6. > > > 2016-02-10 18:07 GMT+01:00 Ken Rice >: > >> I doubt that was ever an intended feature... why not just send the A leg >> out and park the bleg just reverse your notation there >> >> On Wed, Feb 10, 2016 at 10:48 AM, Thomas L?cke > > wrote: >> >>> Hi all, >>> >>> Using 1.6.2 this works from fs_cli and dialplan: >>> >>> originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) >>> >>> Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. >>> >> -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/9cd4ac61/attachment.html From jjserranor at gmail.com Wed Feb 10 22:38:42 2016 From: jjserranor at gmail.com (Jose Serrano) Date: Wed, 10 Feb 2016 20:38:42 +0100 Subject: [Freeswitch-users] P-asserted-identity In-Reply-To: References: Message-ID: Thanks both for your answer. However I tried to do what royj suggested me and still the same behaviour ( freeswith send Remote-Party-ID and not P-asserted-id) I have found a old bug that match exactly with my issue: https://freeswitch.org/jira/browse/FS-527. Yes!, it is closed, but is reproducing to me with my current version of freeswitch 1.4.20 what do you think about it? 2016-02-10 16:56 GMT+01:00 Michael Jerris : > I think the other response answers your question, however I wanted to > clarify one thing. FreeSWITCH is a B2BUA. It is not "replacing" the PAID > with RPID, its a new call that is sent with its own configuration. > > Mike > > > On Feb 10, 2016, at 4:31 AM, Jose Serrano wrote: > > > > Hello. > > > > My freeswitch by default replace the P-asserted-identity by > Remote-Party-ID when routing calls to the gateway. > > I want to send the P-asserted-identity and not the Remote-Party-ID and > for that I tried the following: > > > > I have configured in the outbound gateway definition the following > parameters: > > or > > > > but te behavior is the same. > > > > The only thing that works is to set up the "sip_cid_type" variable in > the dialplan like this: > > data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/> > > > > Anyone knows how I can send the P-Asserted-identity without having to > modify all my dial plan adding the sip_cid_type? > > > > Thanks in avanced > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/b3113567/attachment.html From shafeeq.v at gmail.com Wed Feb 10 23:59:07 2016 From: shafeeq.v at gmail.com (mohammed shafeeque) Date: Thu, 11 Feb 2016 02:29:07 +0530 Subject: [Freeswitch-users] Oneway audio issues in freeswitch Message-ID: Hello All We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine) We have tried multiple softphones and the result is same. Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. Any help or hint will be much appreciated. Thank you, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/6930ca34/attachment.html From italo at freeswitch.org Thu Feb 11 04:54:06 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 10 Feb 2016 22:54:06 -0300 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: You need to look at the sip signaling to see what's going on On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque wrote: > Hello All > > We are getting one way audio issues with some softphones and grandstream > phones behind nat registerd to our freeswitch server. > > Here is scenario: > Grandstream call any extensions (one way audio) > Any extension call Grandstream ( Audio works just fine) > > We have tried multiple softphones and the result is same. > > Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF > issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started > with an upgrade to freeswitch. > > Any help or hint will be much appreciated. > > Thank you, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/63b0db21/attachment-0001.html From jprangi at gmail.com Thu Feb 11 06:45:30 2016 From: jprangi at gmail.com (Jai Rangi) Date: Wed, 10 Feb 2016 19:45:30 -0800 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: We have been looking at that all day, but cant figure out the issue. Funny thing is that its happening only when GS Originate the call. May be we are over looking something. Here are two call example. IPs are modified for security. FreeSWITCH (Version 1.6.6 64bit) is ready freeswitch at internal> sofia_contact 1276 at domain.company.com sofia/internal/sip:1276 at 192.168.1.168:12113 ;fs_nat=yes;fs_path=sip%3A1276%4068.5.194.163%3A12113 freeswitch at internal> sofia_contact 142 at domain.company.com sofia/internal/sip:142 at 172.16.42.13:11852 ;fs_nat=yes;fs_path=sip%3A142%4074.67.200.39%3A33812 142 calls 1276 (1276 does not hear anything) (Seems freeswitch not handling nat properly) On TCP dump, I can see free switching receiving the RTP packet, but trying to deliver to local IP for 1276. 1276 calls 142 (All good both parties can hear) 142 call PSTN number (All good) 1272 call PSTN number (All good) Same configuration, same dialplan works just fine with 1.6.2 and 1.4.18. 1.6.2 had intermittent DTMF issue, we upgraded to 1.6.5, found this one way audio, upgraded to 1.6.6. We have narrowed it down to Grand Stream Softphone and Grad Stream IP phones. Here is trace for Both calls. ###################################### U 74.67.200.39:33812 -> 222.222.222.222:5060 INVITE sip:1276 at domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport. From: "142" ;tag=479799221. To: . Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 140 INVITE. Contact: "142" . Max-Forwards: 70. User-Agent: Grandstream Wave/IOS 1.0.19. Privacy: none. P-Preferred-Identity: "142" . Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 50476 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: . Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 140 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: ;tag=Kt2jU0QN57N8F. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 140 INVITE. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", algorithm=MD5, qop="auth". Content-Length: 0. . ## U 74.67.200.39:33812 -> 222.222.222.222:5060 ACK sip:1276 at domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport. From: "142" ;tag=479799221. To: ;tag=Kt2jU0QN57N8F. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 140 ACK. Content-Length: 0. . # U 74.67.200.39:33812 -> 222.222.222.222:5060 INVITE sip:1276 at domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport. From: "142" ;tag=479799221. To: . Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: "142" . Proxy-Authorization: Digest username="xxxxx", realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri=" sip:1276 at domain.company.com:5060", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003. Max-Forwards: 70. User-Agent: Grandstream Wave/IOS 1.0.19. Privacy: none. P-Preferred-Identity: "142" . Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 50476 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: . Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 141 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 INVITE sip:1276 at 192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKX5H9FDZX1eyNN. Route: . Max-Forwards: 68. From: "Softphone" ;tag=NcN4ypSvZS2DQ. To: . Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Contact: . User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455142324 1455142325 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 17642 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. # # U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 100 Trying. From: Softphone;tag=NcN4ypSvZS2DQ. To: sip:1276 at 192.168.1.168:12113. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Content-Length: 0. . ### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 180 Ringing. From: Softphone;tag=NcN4ypSvZS2DQ. To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Contact: ATAPHONE. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: ;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: . User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 222. P-Asserted-Identity: "1276" . . v=0. o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 27910 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. ############## ## ####### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: Softphone;tag=NcN4ypSvZS2DQ. To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Contact: ATAPHONE. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Content-Type: application/sdp. Content-Length: 207. . v=0. o=1276 87748 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8000 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 ACK sip:1276 at 192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKZQ4tK304U0aUc. Max-Forwards: 70. From: "Softphone" ;tag=NcN4ypSvZS2DQ. To: ;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 ACK. Contact: . Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: ;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: . User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 222. P-Asserted-Identity: "Outbound Call" . . v=0. o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 27910 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. ## U 74.67.200.39:33812 -> 222.222.222.222:5060 ACK sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1657841580;rport. From: "142" ;tag=479799221. To: ;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 141 ACK. Contact: . Proxy-Authorization: Digest username="xxxx", realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri=" sip:1276 at domain.company.com:5060", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003. Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . ############################################################### #### # # ######### ################################################################################################ U 74.67.200.39:33812 -> 222.222.222.222:5060 BYE sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport. From: "142" ;tag=479799221. To: ;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 142 BYE. Contact: . Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK326133779;rport=33812;received=74.67.200.39. From: "142" ;tag=479799221. To: ;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15 at BHC.BG.EC.BD. CSeq: 142 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Content-Length: 0. . ### U 222.222.222.222:5060 -> 68.5.194.163:12113 BYE sip:1276 at 192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK2jg5rmKFKUDKF. Max-Forwards: 70. From: "Softphone" ;tag=NcN4ypSvZS2DQ. To: ;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243920 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . ### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: Softphone;tag=NcN4ypSvZS2DQ. To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243920 BYE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK2jg5rmKFKUDKF. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . ####################################################### ######### # Successful Call with 2 way audio :1276 calls 142 # # # ################################## U 68.5.194.163:12113 -> 222.222.222.222:5060 INVITE sip:142 at domain.company.com SIP/2.0. From: ATAPHONE;tag=VxLf2-dmKe50. To: sip:142 at domain.company.com. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 176 INVITE. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH. Contact: ATAPHONE. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Content-Type: application/sdp. Content-Length: 257. . v=0. o=1276 87384 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8002 RTP/AVP 0 18 8 96. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163. From: ATAPHONE ;tag=VxLf2-dmKe50. To: . Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 176 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163. From: ATAPHONE ;tag=VxLf2-dmKe50. To: ;tag=SjctQ6jcUQD5a. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 176 INVITE. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="domain.company.com", nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb", algorithm=MD5, qop="auth". Content-Length: 0. . # U 68.5.194.163:12113 -> 222.222.222.222:5060 ACK sip:142 at domain.company.com SIP/2.0. From: ATAPHONE;tag=VxLf2-dmKe50. To: sip:142 at domain.company.com;tag=SjctQ6jcUQD5a. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 176 ACK. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH. Content-Length: 0. . # U 68.5.194.163:12113 -> 222.222.222.222:5060 INVITE sip:142 at domain.company.com SIP/2.0. From: ATAPHONE;tag=VxLf2-dmKe50. To: sip:142 at domain.company.com. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 177 INVITE. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0. Contact: ATAPHONE. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Proxy-Authorization: Digest username="yyyy",realm="domain.company.com",uri=" sip:142 at domain.company.com ",response="a87ca168b1869994a6b4782df7bffe99",algorithm=MD5,nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb",qop=auth,cnonce="00017214",nc=00000001. Content-Type: application/sdp. Content-Length: 257. . v=0. o=1276 87384 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8002 RTP/AVP 0 18 8 96. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE ;tag=VxLf2-dmKe50. To: . Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 177 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 INVITE sip:142 at 172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD. Route: . Max-Forwards: 68. From: "ATA" ;tag=U4yBUvmKN9Saj. To: . Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: . User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 18346 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. # U 222.222.222.222:5060 -> 74.67.200.39:33812 INVITE sip:142 at 172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD. Route: . Max-Forwards: 68. From: "ATA" ;tag=U4yBUvmKN9Saj. To: . Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: . User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 18346 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. ## U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" ;tag=U4yBUvmKN9Saj. To: . Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Supported: replaces, path, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" ;tag=U4yBUvmKN9Saj. To: ;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: . Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow-Events: talk, hold. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE ;tag=VxLf2-dmKe50. To: ;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 177 INVITE. Contact: . User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 219. Remote-Party-ID: "142" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 19096 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-16. a=ptime:20. ################################################################## U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" ;tag=U4yBUvmKN9Saj. To: ;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: . Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 26390 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 ACK sip:142 at 172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKB4K487aFSBpUQ. Max-Forwards: 70. From: "ATA" ;tag=U4yBUvmKN9Saj. To: ;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 ACK. Contact: . Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE ;tag=VxLf2-dmKe50. To: ;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 177 INVITE. Contact: . User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 219. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 19096 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-16. a=ptime:20. # U 68.5.194.163:12113 -> 222.222.222.222:5060 ACK sip:142 at 222.222.222.222:5060 SIP/2.0. From: ATAPHONE;tag=VxLf2-dmKe50. To: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 177 ACK. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-VwM5aez*. Contact: ATAPHONE. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . ########################################################################################### ############################## ############################################################################################################################################################## U 74.67.200.39:33812 -> 222.222.222.222:5060 BYE sip:1276 at 222.222.222.222:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport. From: ;tag=1713531477. To: "ATA" ;tag=U4yBUvmKN9Saj. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243933 BYE. Contact: . Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK364522226;rport=33812;received=74.67.200.39. From: ;tag=1713531477. To: "ATA" ;tag=U4yBUvmKN9Saj. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243933 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 BYE sip:1276 at 192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK89gKvrKN5DvSQ. Max-Forwards: 70. From: ;tag=tU5jS13Fr03Qp. To: ATAPHONE ;tag=VxLf2-dmKe50. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 87243943 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. ## U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp. To: ATAPHONE;tag=VxLf2-dmKe50. Call-ID: d44p80-lYfd5f2 at domain.company.com. CSeq: 87243943 BYE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK89gKvrKN5DvSQ. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. On Wed, Feb 10, 2016 at 5:54 PM, ?talo Rossi wrote: > You need to look at the sip signaling to see what's going on > > On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque > wrote: > >> Hello All >> >> We are getting one way audio issues with some softphones and grandstream >> phones behind nat registerd to our freeswitch server. >> >> Here is scenario: >> Grandstream call any extensions (one way audio) >> Any extension call Grandstream ( Audio works just fine) >> >> We have tried multiple softphones and the result is same. >> >> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF >> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started >> with an upgrade to freeswitch. >> >> Any help or hint will be much appreciated. >> >> Thank you, >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ?talo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/ae8bf446/attachment-0001.html From jprangi at gmail.com Thu Feb 11 08:22:55 2016 From: jprangi at gmail.com (Jai Rangi) Date: Wed, 10 Feb 2016 21:22:55 -0800 Subject: [Freeswitch-users] Mod_callcenter ODBC vs File In-Reply-To: References: Message-ID: FYI, Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned state immediately. -Jai On Wed, Jul 22, 2015 at 6:48 PM, Brian West wrote: > Once you test our latest release (1.4.20): > > [image: Inline image 1] > > https://freeswitch.org/jira > > Thanks, > > > On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi wrote: > >> Hello, >> >> FS version: FreeSWITCH Version >> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 >> 18:55:23Z 64bit) >> >> I found a very strange behavior on mod_call center when used with odbc. >> This was easy to reproduce when the agents are busy and new calls come in. >> New member goes in abandoned state immediately and stay in same state >> forever. Even if previous call is complete and Agent is in Waiting state. >> Infact some times new members go in abandoned state immediately even if the >> agents are available. >> >> Everything works perfect once I changed the mod_callcenter.conf.xml >> config to file only mode, new calls go in Waiting state and call is >> connected to agent as soon as the agent is free. >> >> Any idea why the new member goes in abandoned state randomly in place of >> being in waiting state? Can it be due to too small timeout in call center >> module as it query the DB every time? Can that be changed? >> >> Anyone else had smiler problem, any experience or any recommendations >> will be much appreciated. >> >> Thank you, >> -Jai >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/179dce21/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: f0aaba39a4f1b7540df5317cd39fd420.jpg Type: image/jpeg Size: 10741 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/179dce21/attachment.jpg From krice at freeswitch.org Thu Feb 11 09:57:13 2016 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Feb 2016 00:57:13 -0600 Subject: [Freeswitch-users] Mod_callcenter ODBC vs File In-Reply-To: References: Message-ID: is there a jira opened on this? if not, then a dev most likely wont see this On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi wrote: > FYI, > Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned > state immediately. > > -Jai > > > On Wed, Jul 22, 2015 at 6:48 PM, Brian West wrote: > >> Once you test our latest release (1.4.20): >> >> [image: Inline image 1] >> >> https://freeswitch.org/jira >> >> Thanks, >> >> >> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi wrote: >> >>> Hello, >>> >>> FS version: FreeSWITCH Version >>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 >>> 18:55:23Z 64bit) >>> >>> I found a very strange behavior on mod_call center when used with odbc. >>> This was easy to reproduce when the agents are busy and new calls come in. >>> New member goes in abandoned state immediately and stay in same state >>> forever. Even if previous call is complete and Agent is in Waiting state. >>> Infact some times new members go in abandoned state immediately even if the >>> agents are available. >>> >>> Everything works perfect once I changed the mod_callcenter.conf.xml >>> config to file only mode, new calls go in Waiting state and call is >>> connected to agent as soon as the agent is free. >>> >>> Any idea why the new member goes in abandoned state randomly in place of >>> being in waiting state? Can it be due to too small timeout in call center >>> module as it query the DB every time? Can that be changed? >>> >>> Anyone else had smiler problem, any experience or any recommendations >>> will be much appreciated. >>> >>> Thank you, >>> -Jai >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c0bc6cfa/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: f0aaba39a4f1b7540df5317cd39fd420.jpg Type: image/jpeg Size: 10741 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c0bc6cfa/attachment-0001.jpg From jprangi at didforsale.com Thu Feb 11 10:07:20 2016 From: jprangi at didforsale.com (Jai Rangi) Date: Wed, 10 Feb 2016 23:07:20 -0800 Subject: [Freeswitch-users] Mod_callcenter ODBC vs File In-Reply-To: References: Message-ID: I can open Jira for that. It might take a day or two. Right now dealing with another NAT/RTP issue we encountered with the upgrade. Thank you for your response. -Jai *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Wed, Feb 10, 2016 at 10:57 PM, Ken Rice wrote: > is there a jira opened on this? if not, then a dev most likely wont see > this > > > On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi wrote: > >> FYI, >> Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned >> state immediately. >> >> -Jai >> >> >> On Wed, Jul 22, 2015 at 6:48 PM, Brian West wrote: >> >>> Once you test our latest release (1.4.20): >>> >>> [image: Inline image 1] >>> >>> https://freeswitch.org/jira >>> >>> Thanks, >>> >>> >>> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi wrote: >>> >>>> Hello, >>>> >>>> FS version: FreeSWITCH Version >>>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 >>>> 18:55:23Z 64bit) >>>> >>>> I found a very strange behavior on mod_call center when used with odbc. >>>> This was easy to reproduce when the agents are busy and new calls come in. >>>> New member goes in abandoned state immediately and stay in same state >>>> forever. Even if previous call is complete and Agent is in Waiting state. >>>> Infact some times new members go in abandoned state immediately even if the >>>> agents are available. >>>> >>>> Everything works perfect once I changed the mod_callcenter.conf.xml >>>> config to file only mode, new calls go in Waiting state and call is >>>> connected to agent as soon as the agent is free. >>>> >>>> Any idea why the new member goes in abandoned state randomly in place >>>> of being in waiting state? Can it be due to too small timeout in call >>>> center module as it query the DB every time? Can that be changed? >>>> >>>> Anyone else had smiler problem, any experience or any recommendations >>>> will be much appreciated. >>>> >>>> Thank you, >>>> -Jai >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/91454a46/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: f0aaba39a4f1b7540df5317cd39fd420.jpg Type: image/jpeg Size: 10741 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/91454a46/attachment-0001.jpg From bilaln018 at gmail.com Thu Feb 11 12:19:16 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 11 Feb 2016 14:19:16 +0500 Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through JsSIP] Message-ID: Hi all, Currently i am facing an CODEC NEGOTIATION ERROR while calling through JsSIP Caller 1001 Callie 1000, Please view the logs, https://pastebin.freeswitch.org/24550 I have enabled PCMA and PCMU in my var.xml, Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/f621aa83/attachment.html From davidwaf at gmail.com Thu Feb 11 12:46:06 2016 From: davidwaf at gmail.com (David Wafula) Date: Thu, 11 Feb 2016 11:46:06 +0200 Subject: [Freeswitch-users] Two Users Registered ..But calls only going one way Message-ID: Hi all, I have two users who registered in the same domain: user A and B. A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here: http://pastebin.com/gWrrS4zw Am not sure what is causing it. Regards -- David W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/e6baf913/attachment.html From steveayre at gmail.com Thu Feb 11 13:18:18 2016 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Feb 2016 10:18:18 +0000 Subject: [Freeswitch-users] P-asserted-identity In-Reply-To: References: Message-ID: > > The only thing that works is to set up the "sip_cid_type" variable in the > dialplan like this: Try setting sip_cid_type=pid as a variable within the gateway definition, so it's automatically set on every outbound call through that gateway. https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration#GatewaysConfiguration-Variables On 10 February 2016 at 10:31, Jose Serrano wrote: > Hello. > > My freeswitch by default replace the P-asserted-identity by > Remote-Party-ID when routing calls to the gateway. > I want to send the P-asserted-identity and not the Remote-Party-ID and for > that I tried the following: > > I have configured in the outbound gateway definition the following > parameters: > or > > but te behavior is the same. > > The only thing that works is to set up the "sip_cid_type" variable in the > dialplan like this: > data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/> > > Anyone knows how I can send the P-Asserted-identity without having to > modify all my dial plan adding the sip_cid_type? > > Thanks in avanced > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/e0dbdc68/attachment.html From deforceczt at gmail.com Thu Feb 11 15:16:02 2016 From: deforceczt at gmail.com (Vladislav Ivanov) Date: Thu, 11 Feb 2016 14:16:02 +0200 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: <56B0C9B7.8000106@mst.edu> References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: Hey guys, It is true, I haven't ran it as root, I fixed it. But I still have issues with passing more than 50cps, same is happening... 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 This at 25 cps (I bridge the call, so cps*2) Then I add 5 cps more: load average: 1061.40, 333.52 And all OS start's lagging as hell, and i'm unable to find issue, have no idea what is happening... My settings: http://pastebin.com/62B45z4i Anything I'm missing? Really strange that CPU drops like that... 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : > "Run as" != "Start as" > > If you insist on not starting FS as root to let it change user, like most > other daemons/services, you'll have to jump > through a bunch of extra steps using file system capabilities to give it > the ability to set scheduler parameters/etc > that are restricted to root normally. > > -- Nathan > > On 02/02/2016 09:10 AM, Peter Steinbach wrote: > > I've just stumpled over this: > > >Is FreeSWITCH starting with root permissions? It needs this in order > to use the FIFO scheduler and access realtime > > threads. If not started as root, this would explain your CPS limitations. > > > > We like to run Freeswitch as a non privileged user, due to security > concerns. So there are drawbacks here compared to > > running FS as root? Can we somehow quantify the differences? > > > > Best regards > > Peter > > > > > > On 02/02/16 13:58, Bote Man wrote: > >> > >> Is FreeSWITCH starting with root permissions? It needs this in order to > use the FIFO scheduler and access realtime > >> threads. If not started as root, this would explain your CPS > limitations. There are also limits that can be set in the > >> config files. > >> > >> After it starts it drops privileges to those specified on the command > line with ?u and ?g switches. > >> > >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe it > is showing the multiple threads? > >> > >> top ?H shows each thread. > >> > >> --- > >> > >> Bote > >> > >> FreeSWITCH Docs Janitor > >> > >> http://freeswitch.org/confluence > >> > >> *From:*Vladislav Ivanov > >> *Sent:* Tuesday, 02 February, 2016 07:09 > >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? > >> > >> Hey guys, > >> > >> I have a question about freeswitch process/threading usage. > >> So far that I haven't noticed freeswitch to fork himself, I have only 1 > freeswitch instance. > >> http://i.imgur.com/bdbYOwp.png > >> > >> But then I found screenshot of htop with freeswitch and noticed that > there is multiple freeswitch processes being run: > >> http://i.imgur.com/VNpl55z.jpg > >> > >> I'm having issues with "loading" the freeswitch after 50 cps in any > cpu/ram configuration. > >> Be it physical or virtual environment I cant pass the 50 cps mark. > >> I have strange issue with CPU usage on same CPS: > >> > >> http://i.imgur.com/8BdQWVL.png > >> http://i.imgur.com/mWRnoGr.png > >> > >> I timeload test freeswitch with 50cps for 5+ hours, and seems like > there is some kind of leak somewhere. > >> I have tested configuration on: > >> Debian 8 > >> 2 core/8 gb ram > >> 4 core/8 gb ram (graphs are from here) > >> 8 core/32 gb ram > >> > >> and in all the tests I were not able to send more than 50 cps without > CPU dropping to 0 with all system starting to > >> respond really laggy. > >> > >> Test is: > >> sipp -> freeswitch -> sipp > >> > >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan > and 1 profile... > >> Any advice? > >> > >> Thank you all > >> > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > -- > > With kind regards > > Peter Steinbach > > > > Telefaks Services GmbH > > mailto:lists (att) telefaks.de > > Internet:www.telefaks.de > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/de3822c7/attachment-0001.html From gmaruzz at gmail.com Thu Feb 11 16:57:10 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 11 Feb 2016 14:57:10 +0100 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: by default FreeSWITCH has a limit of 30cps... You must edit config files to change that... On Thu, Feb 11, 2016 at 1:16 PM, Vladislav Ivanov wrote: > Hey guys, > > It is true, I haven't ran it as root, I fixed it. > But I still have issues with passing more than 50cps, same is happening... > 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 > This at 25 cps (I bridge the call, so cps*2) > Then I add 5 cps more: > load average: 1061.40, 333.52 > And all OS start's lagging as hell, and i'm unable to find issue, have no > idea what is happening... > > My settings: > http://pastebin.com/62B45z4i > > Anything I'm missing? Really strange that CPU drops like that... > > > > 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : > >> "Run as" != "Start as" >> >> If you insist on not starting FS as root to let it change user, like most >> other daemons/services, you'll have to jump >> through a bunch of extra steps using file system capabilities to give it >> the ability to set scheduler parameters/etc >> that are restricted to root normally. >> >> -- Nathan >> >> On 02/02/2016 09:10 AM, Peter Steinbach wrote: >> > I've just stumpled over this: >> > >Is FreeSWITCH starting with root permissions? It needs this in order >> to use the FIFO scheduler and access realtime >> > threads. If not started as root, this would explain your CPS >> limitations. >> > >> > We like to run Freeswitch as a non privileged user, due to security >> concerns. So there are drawbacks here compared to >> > running FS as root? Can we somehow quantify the differences? >> > >> > Best regards >> > Peter >> > >> > >> > On 02/02/16 13:58, Bote Man wrote: >> >> >> >> Is FreeSWITCH starting with root permissions? It needs this in order >> to use the FIFO scheduler and access realtime >> >> threads. If not started as root, this would explain your CPS >> limitations. There are also limits that can be set in the >> >> config files. >> >> >> >> After it starts it drops privileges to those specified on the command >> line with ?u and ?g switches. >> >> >> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe >> it is showing the multiple threads? >> >> >> >> top ?H shows each thread. >> >> >> >> --- >> >> >> >> Bote >> >> >> >> FreeSWITCH Docs Janitor >> >> >> >> http://freeswitch.org/confluence >> >> >> >> *From:*Vladislav Ivanov >> >> *Sent:* Tuesday, 02 February, 2016 07:09 >> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >> >> >> >> Hey guys, >> >> >> >> I have a question about freeswitch process/threading usage. >> >> So far that I haven't noticed freeswitch to fork himself, I have only >> 1 freeswitch instance. >> >> http://i.imgur.com/bdbYOwp.png >> >> >> >> But then I found screenshot of htop with freeswitch and noticed that >> there is multiple freeswitch processes being run: >> >> http://i.imgur.com/VNpl55z.jpg >> >> >> >> I'm having issues with "loading" the freeswitch after 50 cps in any >> cpu/ram configuration. >> >> Be it physical or virtual environment I cant pass the 50 cps mark. >> >> I have strange issue with CPU usage on same CPS: >> >> >> >> http://i.imgur.com/8BdQWVL.png >> >> http://i.imgur.com/mWRnoGr.png >> >> >> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like >> there is some kind of leak somewhere. >> >> I have tested configuration on: >> >> Debian 8 >> >> 2 core/8 gb ram >> >> 4 core/8 gb ram (graphs are from here) >> >> 8 core/32 gb ram >> >> >> >> and in all the tests I were not able to send more than 50 cps without >> CPU dropping to 0 with all system starting to >> >> respond really laggy. >> >> >> >> Test is: >> >> sipp -> freeswitch -> sipp >> >> >> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan >> and 1 profile... >> >> Any advice? >> >> >> >> Thank you all >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > -- >> > With kind regards >> > Peter Steinbach >> > >> > Telefaks Services GmbH >> > mailto:lists (att) telefaks.de >> > Internet:www.telefaks.de >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/ea8abacf/attachment.html From luis.daniel.lucio at gmail.com Thu Feb 11 05:14:39 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 10 Feb 2016 21:14:39 -0500 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: As a rule of dumb, try turning on rport Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" a ?crit : > You need to look at the sip signaling to see what's going on > > On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque > wrote: > >> Hello All >> >> We are getting one way audio issues with some softphones and grandstream >> phones behind nat registerd to our freeswitch server. >> >> Here is scenario: >> Grandstream call any extensions (one way audio) >> Any extension call Grandstream ( Audio works just fine) >> >> We have tried multiple softphones and the result is same. >> >> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF >> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started >> with an upgrade to freeswitch. >> >> Any help or hint will be much appreciated. >> >> Thank you, >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ?talo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160210/51defab0/attachment-0001.html From mailings at interloop-software.de Thu Feb 11 16:54:02 2016 From: mailings at interloop-software.de (Dominik Steinbrecher) Date: Thu, 11 Feb 2016 14:54:02 +0100 Subject: [Freeswitch-users] Jessie: Unable to locate package Message-ID: Hi, I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie. I followed the guide from: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie When I execute apt-get install I got the following errors: root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all freeswitch-all-dbg gdb Hit http://security.debian.org jessie/updates InRelease Ign http://ftp.de.debian.org jessie InRelease Hit http://ftp.de.debian.org jessie-updates InRelease Hit http://ftp.de.debian.org jessie Release.gpg Hit http://security.debian.org jessie/updates/main Sources Hit http://ftp.de.debian.org jessie Release Hit http://security.debian.org jessie/updates/main i386 Packages Hit http://security.debian.org jessie/updates/main Translation-en Hit http://ftp.de.debian.org jessie-updates/main Sources Get:1 http://ftp.de.debian.org jessie-updates/main i386 Packages/DiffIndex [367 B] Hit http://ftp.de.debian.org jessie-updates/main Translation-en Hit http://ftp.de.debian.org jessie/main Sources Hit http://ftp.de.debian.org jessie/main i386 Packages Hit http://ftp.de.debian.org jessie/main Translation-en Hit http://files.freeswitch.org jessie InRelease Hit http://files.freeswitch.org jessie/main i386 Packages Ign http://files.freeswitch.org jessie/main Translation-en_US Ign http://files.freeswitch.org jessie/main Translation-en Fetched 367 B in 3s (102 B/s) Reading package lists... Done Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package freeswitch-all E: Unable to locate package freeswitch-all-dbg I searched for the freeswitch-all package with aptitude but there?s no such package. Any ideas, what I am doing wrong? Is there anything I could try or check? The debian jessie 8.3 64bit is running as VM under VirtualBox. The host computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just for testing freeswitch. Thanks a lot Dominik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/6d6f153e/attachment.html From gmaruzz at gmail.com Thu Feb 11 17:20:20 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 11 Feb 2016 15:20:20 +0100 Subject: [Freeswitch-users] Jessie: Unable to locate package In-Reply-To: References: Message-ID: are you using an i386 VM? You better use a 64bit VM... (not sure 32 bit is supported) On Thu, Feb 11, 2016 at 2:54 PM, Dominik Steinbrecher < mailings at interloop-software.de> wrote: > Hi, > > I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie. > > I followed the guide from: > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie > > When I execute apt-get install I got the following errors: > > root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all > freeswitch-all-dbg gdb > Hit http://security.debian.org jessie/updates InRelease > Ign http://ftp.de.debian.org jessie InRelease > > Hit http://ftp.de.debian.org jessie-updates InRelease > > Hit http://ftp.de.debian.org jessie Release.gpg > > Hit http://security.debian.org jessie/updates/main Sources > > Hit http://ftp.de.debian.org jessie Release > > Hit http://security.debian.org jessie/updates/main i386 Packages > > Hit http://security.debian.org jessie/updates/main Translation-en > > Hit http://ftp.de.debian.org jessie-updates/main Sources > > Get:1 http://ftp.de.debian.org jessie-updates/main i386 > Packages/DiffIndex [367 B] > Hit http://ftp.de.debian.org jessie-updates/main Translation-en > Hit http://ftp.de.debian.org jessie/main Sources > Hit http://ftp.de.debian.org jessie/main i386 Packages > Hit http://ftp.de.debian.org jessie/main Translation-en > Hit http://files.freeswitch.org jessie InRelease > Hit http://files.freeswitch.org jessie/main i386 Packages > Ign http://files.freeswitch.org jessie/main Translation-en_US > Ign http://files.freeswitch.org jessie/main Translation-en > Fetched 367 B in 3s (102 B/s) > Reading package lists... Done > Reading package lists... Done > Building dependency tree > Reading state information... Done > E: Unable to locate package freeswitch-all > E: Unable to locate package freeswitch-all-dbg > > I searched for the freeswitch-all package with aptitude but there?s no > such package. > > Any ideas, what I am doing wrong? Is there anything I could try or check? > > The debian jessie 8.3 64bit is running as VM under VirtualBox. The host > computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just > for testing freeswitch. > > Thanks a lot > Dominik > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c9d57452/attachment.html From regis.freeswitch.org at tornad.net Thu Feb 11 17:32:13 2016 From: regis.freeswitch.org at tornad.net (Regis M) Date: Thu, 11 Feb 2016 15:32:13 +0100 Subject: [Freeswitch-users] Jessie: Unable to locate package In-Reply-To: References: Message-ID: I confirm, I install 4 or 5 of them each day since 3 days for deployement tests and I have not problem with package 1.6 Debian 8.3 x64 Regard, 2016-02-11 15:20 GMT+01:00 Giovanni Maruzzelli : > are you using an i386 VM? > > You better use a 64bit VM... (not sure 32 bit is supported) > > > On Thu, Feb 11, 2016 at 2:54 PM, Dominik Steinbrecher < > mailings at interloop-software.de> wrote: > >> Hi, >> >> I want to install freeswitch 1.6 on a freshly installed Debian 8.3 >> jessie. >> >> I followed the guide from: >> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie >> >> When I execute apt-get install I got the following errors: >> >> root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all >> freeswitch-all-dbg gdb >> Hit http://security.debian.org jessie/updates InRelease >> Ign http://ftp.de.debian.org jessie InRelease >> >> Hit http://ftp.de.debian.org jessie-updates InRelease >> >> Hit http://ftp.de.debian.org jessie Release.gpg >> >> Hit http://security.debian.org jessie/updates/main Sources >> >> Hit http://ftp.de.debian.org jessie Release >> >> Hit http://security.debian.org jessie/updates/main i386 Packages >> >> Hit http://security.debian.org jessie/updates/main Translation-en >> >> Hit http://ftp.de.debian.org jessie-updates/main Sources >> >> Get:1 http://ftp.de.debian.org jessie-updates/main i386 >> Packages/DiffIndex [367 B] >> Hit http://ftp.de.debian.org jessie-updates/main Translation-en >> Hit http://ftp.de.debian.org jessie/main Sources >> Hit http://ftp.de.debian.org jessie/main i386 Packages >> Hit http://ftp.de.debian.org jessie/main Translation-en >> Hit http://files.freeswitch.org jessie InRelease >> Hit http://files.freeswitch.org jessie/main i386 Packages >> Ign http://files.freeswitch.org jessie/main Translation-en_US >> Ign http://files.freeswitch.org jessie/main Translation-en >> Fetched 367 B in 3s (102 B/s) >> Reading package lists... Done >> Reading package lists... Done >> Building dependency tree >> Reading state information... Done >> E: Unable to locate package freeswitch-all >> E: Unable to locate package freeswitch-all-dbg >> >> I searched for the freeswitch-all package with aptitude but there?s no >> such package. >> >> Any ideas, what I am doing wrong? Is there anything I could try or check? >> >> The debian jessie 8.3 64bit is running as VM under VirtualBox. The host >> computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just >> for testing freeswitch. >> >> Thanks a lot >> Dominik >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/25de5680/attachment-0001.html From brian at freeswitch.org Thu Feb 11 17:37:45 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Feb 2016 08:37:45 -0600 Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through JsSIP] In-Reply-To: References: Message-ID: NO candidate ACL defined, Defaulting to wan.auto sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found. On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi wrote: > Hi all, > > Currently i am facing an CODEC NEGOTIATION ERROR while calling through > JsSIP Caller 1001 Callie 1000, > Please view the logs, > https://pastebin.freeswitch.org/24550 > > I have enabled PCMA and PCMU in my var.xml, > > Regards > Abbasi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/9a56e42a/attachment.html From brian at freeswitch.org Thu Feb 11 17:38:39 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Feb 2016 08:38:39 -0600 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: I sense that your load testing method is flawed and you're just DDoSing the box. On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov wrote: > Hey guys, > > It is true, I haven't ran it as root, I fixed it. > But I still have issues with passing more than 50cps, same is happening... > 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 > This at 25 cps (I bridge the call, so cps*2) > Then I add 5 cps more: > load average: 1061.40, 333.52 > And all OS start's lagging as hell, and i'm unable to find issue, have no > idea what is happening... > > My settings: > http://pastebin.com/62B45z4i > > Anything I'm missing? Really strange that CPU drops like that... > > > > 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : > >> "Run as" != "Start as" >> >> If you insist on not starting FS as root to let it change user, like most >> other daemons/services, you'll have to jump >> through a bunch of extra steps using file system capabilities to give it >> the ability to set scheduler parameters/etc >> that are restricted to root normally. >> >> -- Nathan >> >> On 02/02/2016 09:10 AM, Peter Steinbach wrote: >> > I've just stumpled over this: >> > >Is FreeSWITCH starting with root permissions? It needs this in order >> to use the FIFO scheduler and access realtime >> > threads. If not started as root, this would explain your CPS >> limitations. >> > >> > We like to run Freeswitch as a non privileged user, due to security >> concerns. So there are drawbacks here compared to >> > running FS as root? Can we somehow quantify the differences? >> > >> > Best regards >> > Peter >> > >> > >> > On 02/02/16 13:58, Bote Man wrote: >> >> >> >> Is FreeSWITCH starting with root permissions? It needs this in order >> to use the FIFO scheduler and access realtime >> >> threads. If not started as root, this would explain your CPS >> limitations. There are also limits that can be set in the >> >> config files. >> >> >> >> After it starts it drops privileges to those specified on the command >> line with ?u and ?g switches. >> >> >> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe >> it is showing the multiple threads? >> >> >> >> top ?H shows each thread. >> >> >> >> --- >> >> >> >> Bote >> >> >> >> FreeSWITCH Docs Janitor >> >> >> >> http://freeswitch.org/confluence >> >> >> >> *From:*Vladislav Ivanov >> >> *Sent:* Tuesday, 02 February, 2016 07:09 >> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >> >> >> >> Hey guys, >> >> >> >> I have a question about freeswitch process/threading usage. >> >> So far that I haven't noticed freeswitch to fork himself, I have only >> 1 freeswitch instance. >> >> http://i.imgur.com/bdbYOwp.png >> >> >> >> But then I found screenshot of htop with freeswitch and noticed that >> there is multiple freeswitch processes being run: >> >> http://i.imgur.com/VNpl55z.jpg >> >> >> >> I'm having issues with "loading" the freeswitch after 50 cps in any >> cpu/ram configuration. >> >> Be it physical or virtual environment I cant pass the 50 cps mark. >> >> I have strange issue with CPU usage on same CPS: >> >> >> >> http://i.imgur.com/8BdQWVL.png >> >> http://i.imgur.com/mWRnoGr.png >> >> >> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like >> there is some kind of leak somewhere. >> >> I have tested configuration on: >> >> Debian 8 >> >> 2 core/8 gb ram >> >> 4 core/8 gb ram (graphs are from here) >> >> 8 core/32 gb ram >> >> >> >> and in all the tests I were not able to send more than 50 cps without >> CPU dropping to 0 with all system starting to >> >> respond really laggy. >> >> >> >> Test is: >> >> sipp -> freeswitch -> sipp >> >> >> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan >> and 1 profile... >> >> Any advice? >> >> >> >> Thank you all >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > -- >> > With kind regards >> > Peter Steinbach >> > >> > Telefaks Services GmbH >> > mailto:lists (att) telefaks.de >> > Internet:www.telefaks.de >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/7f1110fd/attachment-0001.html From italo at freeswitch.org Thu Feb 11 17:38:53 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 11 Feb 2016 11:38:53 -0300 Subject: [Freeswitch-users] Mod_callcenter ODBC vs File In-Reply-To: References: Message-ID: Give me a debug log, your odbc settings and callcenter configs so I can take a quick look On Thu, Feb 11, 2016 at 4:07 AM, Jai Rangi wrote: > I can open Jira for that. It might take a day or two. Right now dealing > with another NAT/RTP issue we encountered with the upgrade. > Thank you for your response. > -Jai > > *Jai Rangi* > Cebod Technologies LLC dba DIDforSale/Cebod Telecom > O 949-471-0102 | C 949-419-7634 | F > 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | > www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | > > > > > > > On Wed, Feb 10, 2016 at 10:57 PM, Ken Rice wrote: > >> is there a jira opened on this? if not, then a dev most likely wont see >> this >> >> >> On Wed, Feb 10, 2016 at 11:22 PM, Jai Rangi wrote: >> >>> FYI, >>> Tested with 1.6.2, 1.6.5. Still the same issue. New call go to abandoned >>> state immediately. >>> >>> -Jai >>> >>> >>> On Wed, Jul 22, 2015 at 6:48 PM, Brian West >>> wrote: >>> >>>> Once you test our latest release (1.4.20): >>>> >>>> [image: Inline image 1] >>>> >>>> https://freeswitch.org/jira >>>> >>>> Thanks, >>>> >>>> >>>> On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi wrote: >>>> >>>>> Hello, >>>>> >>>>> FS version: FreeSWITCH Version >>>>> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 >>>>> 18:55:23Z 64bit) >>>>> >>>>> I found a very strange behavior on mod_call center when used with >>>>> odbc. This was easy to reproduce when the agents are busy and new calls >>>>> come in. New member goes in abandoned state immediately and stay in same >>>>> state forever. Even if previous call is complete and Agent is in Waiting >>>>> state. Infact some times new members go in abandoned state immediately even >>>>> if the agents are available. >>>>> >>>>> Everything works perfect once I changed the mod_callcenter.conf.xml >>>>> config to file only mode, new calls go in Waiting state and call is >>>>> connected to agent as soon as the agent is free. >>>>> >>>>> Any idea why the new member goes in abandoned state randomly in place >>>>> of being in waiting state? Can it be due to too small timeout in call >>>>> center module as it query the DB every time? Can that be changed? >>>>> >>>>> Anyone else had smiler problem, any experience or any recommendations >>>>> will be much appreciated. >>>>> >>>>> Thank you, >>>>> -Jai >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/f1c7a09b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: f0aaba39a4f1b7540df5317cd39fd420.jpg Type: image/jpeg Size: 10741 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/f1c7a09b/attachment-0001.jpg From deforceczt at gmail.com Thu Feb 11 17:45:24 2016 From: deforceczt at gmail.com (Vladislav Ivanov) Date: Thu, 11 Feb 2016 16:45:24 +0200 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: I have changed it: If there was limit - there would be no CPU hogging. Here is status command: FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready 0 session(s) since startup 0 session(s) - peak 0, last 5min 0 0 session(s) per Sec out of max 400, peak 0, last 5min 0 10000 session(s) max min idle cpu 0.00/100.00 Current Stack Size/Max 240K/8192K I can believe that i'm just DDoSing my box. I'm using simple uac/uas from sipp, and certain load is ok, but after that it goes straight down with increase of just 5 cps... I mean it is strange that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded. 2016-02-11 16:38 GMT+02:00 Brian West : > I sense that your load testing method is flawed and you're just DDoSing > the box. > > On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov > wrote: > >> Hey guys, >> >> It is true, I haven't ran it as root, I fixed it. >> But I still have issues with passing more than 50cps, same is >> happening... >> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 >> This at 25 cps (I bridge the call, so cps*2) >> Then I add 5 cps more: >> load average: 1061.40, 333.52 >> And all OS start's lagging as hell, and i'm unable to find issue, have no >> idea what is happening... >> >> My settings: >> http://pastebin.com/62B45z4i >> >> Anything I'm missing? Really strange that CPU drops like that... >> >> >> >> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : >> >>> "Run as" != "Start as" >>> >>> If you insist on not starting FS as root to let it change user, like >>> most other daemons/services, you'll have to jump >>> through a bunch of extra steps using file system capabilities to give it >>> the ability to set scheduler parameters/etc >>> that are restricted to root normally. >>> >>> -- Nathan >>> >>> On 02/02/2016 09:10 AM, Peter Steinbach wrote: >>> > I've just stumpled over this: >>> > >Is FreeSWITCH starting with root permissions? It needs this in order >>> to use the FIFO scheduler and access realtime >>> > threads. If not started as root, this would explain your CPS >>> limitations. >>> > >>> > We like to run Freeswitch as a non privileged user, due to security >>> concerns. So there are drawbacks here compared to >>> > running FS as root? Can we somehow quantify the differences? >>> > >>> > Best regards >>> > Peter >>> > >>> > >>> > On 02/02/16 13:58, Bote Man wrote: >>> >> >>> >> Is FreeSWITCH starting with root permissions? It needs this in order >>> to use the FIFO scheduler and access realtime >>> >> threads. If not started as root, this would explain your CPS >>> limitations. There are also limits that can be set in the >>> >> config files. >>> >> >>> >> After it starts it drops privileges to those specified on the command >>> line with ?u and ?g switches. >>> >> >>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe >>> it is showing the multiple threads? >>> >> >>> >> top ?H shows each thread. >>> >> >>> >> --- >>> >> >>> >> Bote >>> >> >>> >> FreeSWITCH Docs Janitor >>> >> >>> >> http://freeswitch.org/confluence >>> >> >>> >> *From:*Vladislav Ivanov >>> >> *Sent:* Tuesday, 02 February, 2016 07:09 >>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >>> >> >>> >> Hey guys, >>> >> >>> >> I have a question about freeswitch process/threading usage. >>> >> So far that I haven't noticed freeswitch to fork himself, I have only >>> 1 freeswitch instance. >>> >> http://i.imgur.com/bdbYOwp.png >>> >> >>> >> But then I found screenshot of htop with freeswitch and noticed that >>> there is multiple freeswitch processes being run: >>> >> http://i.imgur.com/VNpl55z.jpg >>> >> >>> >> I'm having issues with "loading" the freeswitch after 50 cps in any >>> cpu/ram configuration. >>> >> Be it physical or virtual environment I cant pass the 50 cps mark. >>> >> I have strange issue with CPU usage on same CPS: >>> >> >>> >> http://i.imgur.com/8BdQWVL.png >>> >> http://i.imgur.com/mWRnoGr.png >>> >> >>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like >>> there is some kind of leak somewhere. >>> >> I have tested configuration on: >>> >> Debian 8 >>> >> 2 core/8 gb ram >>> >> 4 core/8 gb ram (graphs are from here) >>> >> 8 core/32 gb ram >>> >> >>> >> and in all the tests I were not able to send more than 50 cps without >>> CPU dropping to 0 with all system starting to >>> >> respond really laggy. >>> >> >>> >> Test is: >>> >> sipp -> freeswitch -> sipp >>> >> >>> >> Just 1 dialpeer with bridge action. No gateways. Just simple dialplan >>> and 1 profile... >>> >> Any advice? >>> >> >>> >> Thank you all >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://confluence.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > -- >>> > With kind regards >>> > Peter Steinbach >>> > >>> > Telefaks Services GmbH >>> > mailto:lists (att) telefaks.de >>> > Internet:www.telefaks.de >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> -- >>> ------------------------------------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/4dba9d36/attachment.html From bilaln018 at gmail.com Thu Feb 11 17:54:05 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 11 Feb 2016 19:54:05 +0500 Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through JsSIP] In-Reply-To: References: Message-ID: Thanks brain for reply but what i need to do, how can i get rid of this warning. many thanks abbasi On Thursday, February 11, 2016, Brian West wrote: > NO candidate ACL defined, Defaulting to wan.auto > sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found. > > On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi > wrote: > >> Hi all, >> >> Currently i am facing an CODEC NEGOTIATION ERROR while calling through >> JsSIP Caller 1001 Callie 1000, >> Please view the logs, >> https://pastebin.freeswitch.org/24550 >> >> I have enabled PCMA and PCMU in my var.xml, >> >> Regards >> Abbasi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/ebd0d684/attachment-0001.html From krice at freeswitch.org Thu Feb 11 17:58:19 2016 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Feb 2016 08:58:19 -0600 Subject: [Freeswitch-users] Jessie: Unable to locate package In-Reply-To: References: Message-ID: are you sure you are using 64bit? show us the contents of your apt sources.list.d file for freeswitch and uname -a On Thu, Feb 11, 2016 at 7:54 AM, Dominik Steinbrecher < mailings at interloop-software.de> wrote: > Hi, > > I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie. > > I followed the guide from: > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie > > When I execute apt-get install I got the following errors: > > root at freeswitch:~# apt-get update && apt-get install -y freeswitch-all > freeswitch-all-dbg gdb > Hit http://security.debian.org jessie/updates InRelease > Ign http://ftp.de.debian.org jessie InRelease > > Hit http://ftp.de.debian.org jessie-updates InRelease > > Hit http://ftp.de.debian.org jessie Release.gpg > > Hit http://security.debian.org jessie/updates/main Sources > > Hit http://ftp.de.debian.org jessie Release > > Hit http://security.debian.org jessie/updates/main i386 Packages > > Hit http://security.debian.org jessie/updates/main Translation-en > > Hit http://ftp.de.debian.org jessie-updates/main Sources > > Get:1 http://ftp.de.debian.org jessie-updates/main i386 > Packages/DiffIndex [367 B] > Hit http://ftp.de.debian.org jessie-updates/main Translation-en > Hit http://ftp.de.debian.org jessie/main Sources > Hit http://ftp.de.debian.org jessie/main i386 Packages > Hit http://ftp.de.debian.org jessie/main Translation-en > Hit http://files.freeswitch.org jessie InRelease > Hit http://files.freeswitch.org jessie/main i386 Packages > Ign http://files.freeswitch.org jessie/main Translation-en_US > Ign http://files.freeswitch.org jessie/main Translation-en > Fetched 367 B in 3s (102 B/s) > Reading package lists... Done > Reading package lists... Done > Building dependency tree > Reading state information... Done > E: Unable to locate package freeswitch-all > E: Unable to locate package freeswitch-all-dbg > > I searched for the freeswitch-all package with aptitude but there?s no > such package. > > Any ideas, what I am doing wrong? Is there anything I could try or check? > > The debian jessie 8.3 64bit is running as VM under VirtualBox. The host > computer is a Mac Mini with OS X 10.11.1. It?s a freshly installed VM just > for testing freeswitch. > > Thanks a lot > Dominik > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/4c3cf043/attachment.html From adnan.ahmed1 at gmail.com Thu Feb 11 17:32:10 2016 From: adnan.ahmed1 at gmail.com (Adnan Ahmed) Date: Thu, 11 Feb 2016 09:32:10 -0500 Subject: [Freeswitch-users] FS Bridged call, no RTP until DTMF pressed Message-ID: Hi, I have a peculiar situation in which I'm hoping someone can help me out with. I have a Dahdi trunk coming into Asterisk (*), which then sends the call directly to freeswitch (FS), FS will then bridge this incoming call to a SIP device. The problem i'm having is that when FS bridges the call there is no media (or RTP packets) sent back to asterisk until I press a dtmf key from the caller side. The reason that * is there is due to the fact that mod_freeTDM for FS wasn't able to configure the trunk parameters required to control the T1 (E&M with Feature Group B MF), with chan_dahdi in * i was able to set that up with signalling=featb. The dialplan in asterisk is as follows, [from-pstn] > exten => _X.,1,NoOp(Incoming DID matches as ${EXTEN}) > exten => _X.,n,Answer() > exten => _X.,n,Set(CALLERID(all)="0000000000"<0000000000>) > exten => _X.,n,Dial(SIP/freeswitch/1819${EXTEN:0:7},90,M(send-dtmf-1)r) > exten => _X.,n,Hangup() > > [macro-send-dtmf-1] > exten => s,1,SendDTMF(1) I tried sending a DTMF from astersk, and FS recognizes the DTMF, but still no RTP until the key is physically pressed on the caller side. The asterisk dialplan is very simple, answer the incoming dahdi call and send it to FS via SIP. Once the DTMF is pressed, the audio is complete and no issues anymore, so its not a routing, or firewall issue. Both asterisk and FS run on the same machine (* on port 5065, and FS on 5060). Looking at the tcpdump traces, there really is no RTP from FS until after the DTMF is pressed, but the RTP from asterisk is always there. I have the output of "sofia global siptrace on" at the following pastebin: https://pastebin.freeswitch.org/24552 In that SIP trace you will see the call as follows, Incoming call from * bridge to SIP device Failure to connect to SIP device Forward call to voicemail bridge to voicemail connects to voicemail system hangup I can press the DTMF at any point once the first bridge is dialed and will start hearing the audio from that point onwards ... in this case i pressed the DTMF key 1 (you see it being recognized in the FS sip trace log). It makes no difference if I wait to press the DTMF till the second bridge or after the second bridge connects. I have even tried it with a sip device that answers on the first bridge session, and its the same scenario: no audio until dtmf is pressed, again making no difference if its pressed right away or 10 seconds after the call is connected and the other party can hear me but i don't hear them until i press the dtmf. Thanks, Adnan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/b637f129/attachment.html From anthony.minessale at gmail.com Thu Feb 11 19:15:31 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Feb 2016 10:15:31 -0600 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: Without specific details about what the CPU is, how many there are and the version of the OS and many other factors its not easy to answer you. We frown on these load questions because, just like now, it results in large threads and many people spending a lot of time trying to understand the parameters. On Thu, Feb 11, 2016 at 8:45 AM, Vladislav Ivanov wrote: > I have changed it: > > > If there was limit - there would be no CPU hogging. > Here is status command: > FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready > 0 session(s) since startup > 0 session(s) - peak 0, last 5min 0 > 0 session(s) per Sec out of max 400, peak 0, last 5min 0 > 10000 session(s) max > min idle cpu 0.00/100.00 > Current Stack Size/Max 240K/8192K > > I can believe that i'm just DDoSing my box. > I'm using simple uac/uas from sipp, and certain load is ok, but after that > it goes straight down with increase of just 5 cps... I mean it is strange > that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded. > > 2016-02-11 16:38 GMT+02:00 Brian West : > >> I sense that your load testing method is flawed and you're just DDoSing >> the box. >> >> On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov >> wrote: >> >>> Hey guys, >>> >>> It is true, I haven't ran it as root, I fixed it. >>> But I still have issues with passing more than 50cps, same is >>> happening... >>> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 >>> This at 25 cps (I bridge the call, so cps*2) >>> Then I add 5 cps more: >>> load average: 1061.40, 333.52 >>> And all OS start's lagging as hell, and i'm unable to find issue, have >>> no idea what is happening... >>> >>> My settings: >>> http://pastebin.com/62B45z4i >>> >>> Anything I'm missing? Really strange that CPU drops like that... >>> >>> >>> >>> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : >>> >>>> "Run as" != "Start as" >>>> >>>> If you insist on not starting FS as root to let it change user, like >>>> most other daemons/services, you'll have to jump >>>> through a bunch of extra steps using file system capabilities to give >>>> it the ability to set scheduler parameters/etc >>>> that are restricted to root normally. >>>> >>>> -- Nathan >>>> >>>> On 02/02/2016 09:10 AM, Peter Steinbach wrote: >>>> > I've just stumpled over this: >>>> > >Is FreeSWITCH starting with root permissions? It needs this in >>>> order to use the FIFO scheduler and access realtime >>>> > threads. If not started as root, this would explain your CPS >>>> limitations. >>>> > >>>> > We like to run Freeswitch as a non privileged user, due to security >>>> concerns. So there are drawbacks here compared to >>>> > running FS as root? Can we somehow quantify the differences? >>>> > >>>> > Best regards >>>> > Peter >>>> > >>>> > >>>> > On 02/02/16 13:58, Bote Man wrote: >>>> >> >>>> >> Is FreeSWITCH starting with root permissions? It needs this in order >>>> to use the FIFO scheduler and access realtime >>>> >> threads. If not started as root, this would explain your CPS >>>> limitations. There are also limits that can be set in the >>>> >> config files. >>>> >> >>>> >> After it starts it drops privileges to those specified on the >>>> command line with ?u and ?g switches. >>>> >> >>>> >> FreeSWITCH uses multi-threading. I do not know about htop, but maybe >>>> it is showing the multiple threads? >>>> >> >>>> >> top ?H shows each thread. >>>> >> >>>> >> --- >>>> >> >>>> >> Bote >>>> >> >>>> >> FreeSWITCH Docs Janitor >>>> >> >>>> >> http://freeswitch.org/confluence >>>> >> >>>> >> *From:*Vladislav Ivanov >>>> >> *Sent:* Tuesday, 02 February, 2016 07:09 >>>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >>>> >> >>>> >> Hey guys, >>>> >> >>>> >> I have a question about freeswitch process/threading usage. >>>> >> So far that I haven't noticed freeswitch to fork himself, I have >>>> only 1 freeswitch instance. >>>> >> http://i.imgur.com/bdbYOwp.png >>>> >> >>>> >> But then I found screenshot of htop with freeswitch and noticed that >>>> there is multiple freeswitch processes being run: >>>> >> http://i.imgur.com/VNpl55z.jpg >>>> >> >>>> >> I'm having issues with "loading" the freeswitch after 50 cps in any >>>> cpu/ram configuration. >>>> >> Be it physical or virtual environment I cant pass the 50 cps mark. >>>> >> I have strange issue with CPU usage on same CPS: >>>> >> >>>> >> http://i.imgur.com/8BdQWVL.png >>>> >> http://i.imgur.com/mWRnoGr.png >>>> >> >>>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like >>>> there is some kind of leak somewhere. >>>> >> I have tested configuration on: >>>> >> Debian 8 >>>> >> 2 core/8 gb ram >>>> >> 4 core/8 gb ram (graphs are from here) >>>> >> 8 core/32 gb ram >>>> >> >>>> >> and in all the tests I were not able to send more than 50 cps >>>> without CPU dropping to 0 with all system starting to >>>> >> respond really laggy. >>>> >> >>>> >> Test is: >>>> >> sipp -> freeswitch -> sipp >>>> >> >>>> >> Just 1 dialpeer with bridge action. No gateways. Just simple >>>> dialplan and 1 profile... >>>> >> Any advice? >>>> >> >>>> >> Thank you all >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://confluence.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > -- >>>> > With kind regards >>>> > Peter Steinbach >>>> > >>>> > Telefaks Services GmbH >>>> > mailto:lists (att) telefaks.de >>>> > Internet:www.telefaks.de >>>> > >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://confluence.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> >>>> -- >>>> ------------------------------------------------------------ >>>> Nathan Neulinger nneul at mst.edu >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/1605f851/attachment-0001.html From gmaruzz at gmail.com Thu Feb 11 19:18:57 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 11 Feb 2016 17:18:57 +0100 Subject: [Freeswitch-users] Does freeswitch forks his processes? In-Reply-To: References: <005d01d15db9$744a3bd0$5cdeb370$@botecomm.com> <56B0C6FE.9070507@telefaks.de> <56B0C9B7.8000106@mst.edu> Message-ID: :)) guys like talk about fast cars and :)) On Thu, Feb 11, 2016 at 5:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Without specific details about what the CPU is, how many there are and the > version of the OS and many other factors its not easy to answer you. > We frown on these load questions because, just like now, it results in > large threads and many people spending a lot of time trying to understand > the parameters. > > > > > On Thu, Feb 11, 2016 at 8:45 AM, Vladislav Ivanov > wrote: > >> I have changed it: >> >> >> If there was limit - there would be no CPU hogging. >> Here is status command: >> FreeSWITCH (Version 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit) is ready >> 0 session(s) since startup >> 0 session(s) - peak 0, last 5min 0 >> 0 session(s) per Sec out of max 400, peak 0, last 5min 0 >> 10000 session(s) max >> min idle cpu 0.00/100.00 >> Current Stack Size/Max 240K/8192K >> >> I can believe that i'm just DDoSing my box. >> I'm using simple uac/uas from sipp, and certain load is ok, but after >> that it goes straight down with increase of just 5 cps... I mean it is >> strange that with 20 cps it's 25% loaded and with 25 cps it's 100% loaded. >> >> 2016-02-11 16:38 GMT+02:00 Brian West : >> >>> I sense that your load testing method is flawed and you're just DDoSing >>> the box. >>> >>> On Thu, Feb 11, 2016 at 6:16 AM, Vladislav Ivanov >>> wrote: >>> >>>> Hey guys, >>>> >>>> It is true, I haven't ran it as root, I fixed it. >>>> But I still have issues with passing more than 50cps, same is >>>> happening... >>>> 4 cores CPU - CPU is good, 80% idle, load average: 0.81, 0.98 >>>> This at 25 cps (I bridge the call, so cps*2) >>>> Then I add 5 cps more: >>>> load average: 1061.40, 333.52 >>>> And all OS start's lagging as hell, and i'm unable to find issue, have >>>> no idea what is happening... >>>> >>>> My settings: >>>> http://pastebin.com/62B45z4i >>>> >>>> Anything I'm missing? Really strange that CPU drops like that... >>>> >>>> >>>> >>>> 2016-02-02 17:22 GMT+02:00 Nathan Neulinger : >>>> >>>>> "Run as" != "Start as" >>>>> >>>>> If you insist on not starting FS as root to let it change user, like >>>>> most other daemons/services, you'll have to jump >>>>> through a bunch of extra steps using file system capabilities to give >>>>> it the ability to set scheduler parameters/etc >>>>> that are restricted to root normally. >>>>> >>>>> -- Nathan >>>>> >>>>> On 02/02/2016 09:10 AM, Peter Steinbach wrote: >>>>> > I've just stumpled over this: >>>>> > >Is FreeSWITCH starting with root permissions? It needs this in >>>>> order to use the FIFO scheduler and access realtime >>>>> > threads. If not started as root, this would explain your CPS >>>>> limitations. >>>>> > >>>>> > We like to run Freeswitch as a non privileged user, due to security >>>>> concerns. So there are drawbacks here compared to >>>>> > running FS as root? Can we somehow quantify the differences? >>>>> > >>>>> > Best regards >>>>> > Peter >>>>> > >>>>> > >>>>> > On 02/02/16 13:58, Bote Man wrote: >>>>> >> >>>>> >> Is FreeSWITCH starting with root permissions? It needs this in >>>>> order to use the FIFO scheduler and access realtime >>>>> >> threads. If not started as root, this would explain your CPS >>>>> limitations. There are also limits that can be set in the >>>>> >> config files. >>>>> >> >>>>> >> After it starts it drops privileges to those specified on the >>>>> command line with ?u and ?g switches. >>>>> >> >>>>> >> FreeSWITCH uses multi-threading. I do not know about htop, but >>>>> maybe it is showing the multiple threads? >>>>> >> >>>>> >> top ?H shows each thread. >>>>> >> >>>>> >> --- >>>>> >> >>>>> >> Bote >>>>> >> >>>>> >> FreeSWITCH Docs Janitor >>>>> >> >>>>> >> http://freeswitch.org/confluence >>>>> >> >>>>> >> *From:*Vladislav Ivanov >>>>> >> *Sent:* Tuesday, 02 February, 2016 07:09 >>>>> >> *Subject:* [Freeswitch-users] Does freeswitch forks his processes? >>>>> >> >>>>> >> Hey guys, >>>>> >> >>>>> >> I have a question about freeswitch process/threading usage. >>>>> >> So far that I haven't noticed freeswitch to fork himself, I have >>>>> only 1 freeswitch instance. >>>>> >> http://i.imgur.com/bdbYOwp.png >>>>> >> >>>>> >> But then I found screenshot of htop with freeswitch and noticed >>>>> that there is multiple freeswitch processes being run: >>>>> >> http://i.imgur.com/VNpl55z.jpg >>>>> >> >>>>> >> I'm having issues with "loading" the freeswitch after 50 cps in any >>>>> cpu/ram configuration. >>>>> >> Be it physical or virtual environment I cant pass the 50 cps mark. >>>>> >> I have strange issue with CPU usage on same CPS: >>>>> >> >>>>> >> http://i.imgur.com/8BdQWVL.png >>>>> >> http://i.imgur.com/mWRnoGr.png >>>>> >> >>>>> >> I timeload test freeswitch with 50cps for 5+ hours, and seems like >>>>> there is some kind of leak somewhere. >>>>> >> I have tested configuration on: >>>>> >> Debian 8 >>>>> >> 2 core/8 gb ram >>>>> >> 4 core/8 gb ram (graphs are from here) >>>>> >> 8 core/32 gb ram >>>>> >> >>>>> >> and in all the tests I were not able to send more than 50 cps >>>>> without CPU dropping to 0 with all system starting to >>>>> >> respond really laggy. >>>>> >> >>>>> >> Test is: >>>>> >> sipp -> freeswitch -> sipp >>>>> >> >>>>> >> Just 1 dialpeer with bridge action. No gateways. Just simple >>>>> dialplan and 1 profile... >>>>> >> Any advice? >>>>> >> >>>>> >> Thank you all >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.com >>>>> >> >>>>> >> Official FreeSWITCH Sites >>>>> >> http://www.freeswitch.org >>>>> >> http://confluence.freeswitch.org >>>>> >> http://www.cluecon.com >>>>> >> >>>>> >> FreeSWITCH-users mailing list >>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > -- >>>>> > With kind regards >>>>> > Peter Steinbach >>>>> > >>>>> > Telefaks Services GmbH >>>>> > mailto:lists (att) telefaks.de >>>>> > Internet:www.telefaks.de >>>>> > >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://confluence.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> >>>>> -- >>>>> ------------------------------------------------------------ >>>>> Nathan Neulinger nneul at mst.edu >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c6b4a4a3/attachment-0001.html From mike at jerris.com Thu Feb 11 19:21:11 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Feb 2016 10:21:11 -0600 Subject: [Freeswitch-users] Two Users Registered ..But calls only going one way In-Reply-To: References: Message-ID: <792D69F0-C82B-4DE7-B7D6-58F19FADC60C@jerris.com> Sip trace would help... is call forwarding turned on on the phone? > On Feb 11, 2016, at 3:46 AM, David Wafula wrote: > > Hi all, > I have two users who registered in the same domain: user A and B. > A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here: > > > http://pastebin.com/gWrrS4zw > > Am not sure what is causing it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/6aed113b/attachment.html From shafeeq.v at gmail.com Thu Feb 11 19:28:45 2016 From: shafeeq.v at gmail.com (mohammed shafeeque) Date: Thu, 11 Feb 2016 21:58:45 +0530 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > As a rule of dumb, try turning on rport > Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" a ?crit : > >> You need to look at the sip signaling to see what's going on >> >> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque >> wrote: >> >>> Hello All >>> >>> We are getting one way audio issues with some softphones and grandstream >>> phones behind nat registerd to our freeswitch server. >>> >>> Here is scenario: >>> Grandstream call any extensions (one way audio) >>> Any extension call Grandstream ( Audio works just fine) >>> >>> We have tried multiple softphones and the result is same. >>> >>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF >>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started >>> with an upgrade to freeswitch. >>> >>> Any help or hint will be much appreciated. >>> >>> Thank you, >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> ?talo Rossi >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/b73548aa/attachment.html From stefano.favaro at edistar.com Thu Feb 11 19:33:10 2016 From: stefano.favaro at edistar.com (Stefano Favaro) Date: Thu, 11 Feb 2016 17:33:10 +0100 (CET) Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: <374662561.6043.1455208118323.JavaMail.root@mailserver.edistar.com> Message-ID: <965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com> Hello, I have a problem with the mod_spy module. It seems that it just plays music and do not actually spy. I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. This is the dialplan: If I dial 881000, for example, It means I want to spy on user 1000. I have in and out calls from user 1000 but I can't hear. userspy_show in fs_cli, I get : 1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684 1 total spy I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) SF. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c2f094cb/attachment.html From mailings at interloop-software.de Thu Feb 11 19:35:33 2016 From: mailings at interloop-software.de (Dominik Steinbrecher) Date: Thu, 11 Feb 2016 17:35:33 +0100 Subject: [Freeswitch-users] Jessie: Unable to locate package In-Reply-To: References: Message-ID: <2D6C19DC-5C1F-4590-B7A4-1D1708F5FB1C@interloop-software.de> Thanks for your answers and help. > show us the contents of your apt sources.list.d file for freeswitch and uname -a root at freeswitch:~# cat /etc/apt/sources.list.d/freeswitch.list deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main root at freeswitch:~# uname -a Linux freeswitch 3.16.0-4-686-pae #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) i686 GNU/Linux > are you sure you are using 64bit? I thought that I was running a 64bit OS, but it looks like I was wrong. As I showed above, uname -a said i686 so its 32bit. Sorry, for the wrong information in my first post. Looks like that I was choosing the wrong netinst-image for download after selecting 64bit OS in VirtualBox. I just installed a new vm with arch amd64 and everything seems to be fine. The freeswitch installation just finished without any errors. So, thank you a lot for your help Dominik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/8f9d857e/attachment.html From krice at freeswitch.org Thu Feb 11 19:36:37 2016 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Feb 2016 10:36:37 -0600 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque wrote: > Surprised that no one else experienced this problem. Can anyone give any > hint. Really Dont want to move back to 1.4.x > > On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> As a rule of dumb, try turning on rport >> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" a ?crit : >> >>> You need to look at the sip signaling to see what's going on >>> >>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque >> > wrote: >>> >>>> Hello All >>>> >>>> We are getting one way audio issues with some softphones and >>>> grandstream phones behind nat registerd to our freeswitch server. >>>> >>>> Here is scenario: >>>> Grandstream call any extensions (one way audio) >>>> Any extension call Grandstream ( Audio works just fine) >>>> >>>> We have tried multiple softphones and the result is same. >>>> >>>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF >>>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started >>>> with an upgrade to freeswitch. >>>> >>>> Any help or hint will be much appreciated. >>>> >>>> Thank you, >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> ?talo Rossi >>> italo at freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/5fd6b028/attachment-0001.html From lists at kavun.ch Thu Feb 11 19:53:55 2016 From: lists at kavun.ch (Emrah) Date: Thu, 11 Feb 2016 17:53:55 +0100 Subject: [Freeswitch-users] High availability on different networks Message-ID: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Hi list, I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks. I am trying to achieve the following: - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries. - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense. - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up. I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions. I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues. I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless. Best, E From jjserranor at gmail.com Thu Feb 11 19:59:25 2016 From: jjserranor at gmail.com (Jose Serrano) Date: Thu, 11 Feb 2016 17:59:25 +0100 Subject: [Freeswitch-users] P-asserted-identity In-Reply-To: References: Message-ID: Hello Sorry. I made a mistake checking what "royj at yandex.ru" told me. I can confirm that using works fine. Nevertheless "royj at yandex.ru" told me to apply it in the incoming profile, but it works when I have applied in the outbound profile. Thanks everybody. I really apreciate your help SOLVED 2016-02-11 11:18 GMT+01:00 Steven Ayre : > The only thing that works is to set up the "sip_cid_type" variable in the >> dialplan like this: > > > > > Try setting sip_cid_type=pid as a variable within the gateway definition, > so it's automatically set on every outbound call through that gateway. > > > https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration#GatewaysConfiguration-Variables > > On 10 February 2016 at 10:31, Jose Serrano wrote: > >> Hello. >> >> My freeswitch by default replace the P-asserted-identity by >> Remote-Party-ID when routing calls to the gateway. >> I want to send the P-asserted-identity and not the Remote-Party-ID and >> for that I tried the following: >> >> I have configured in the outbound gateway definition the following >> parameters: >> or >> >> but te behavior is the same. >> >> The only thing that works is to set up the "sip_cid_type" variable in the >> dialplan like this: >> > data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/> >> >> Anyone knows how I can send the P-Asserted-identity without having to >> modify all my dial plan adding the sip_cid_type? >> >> Thanks in avanced >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/c6d6455d/attachment.html From italo at freeswitch.org Thu Feb 11 20:52:46 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 11 Feb 2016 14:52:46 -0300 Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: <965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com> References: <374662561.6043.1455208118323.JavaMail.root@mailserver.edistar.com> <965414335.6054.1455208390800.JavaMail.root@mailserver.edistar.com> Message-ID: Stefano, Can you post your debug logs (/log 7)? Use https://pastebin.freeswitch.org/ On Thu, Feb 11, 2016 at 1:33 PM, Stefano Favaro wrote: > Hello, > > I have a problem with the mod_spy module. > It seems that it just plays music and do not actually spy. > I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. > Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. > > This is the dialplan: > > > > > > > > > > If I dial 881000, for example, It means I want to spy on user 1000. > I have in and out calls from user 1000 but I can't hear. > userspy_show in fs_cli, I get : > 1000 at myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684 > > 1 total spy > > I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > SF. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/e5069406/attachment.html From ahabiba at gmail.com Thu Feb 11 21:05:58 2016 From: ahabiba at gmail.com (Ahmed Habiba) Date: Thu, 11 Feb 2016 21:05:58 +0300 Subject: [Freeswitch-users] Two Users Registered ..But calls only going one way In-Reply-To: References: Message-ID: Are you using TLS? is GS configures with nat configuration correctly? > > From: Michael Jerris > > Subject: Re: [Freeswitch-users] Two Users Registered ..But calls only going one way > Date: February 11, 2016 at 7:21:11 PM GMT+3 > To: FreeSWITCH Users Help > > Reply-To: FreeSWITCH Users Help > > > > Sip trace would help... is call forwarding turned on on the phone? > >> On Feb 11, 2016, at 3:46 AM, David Wafula > wrote: >> >> Hi all, >> I have two users who registered in the same domain: user A and B. >> A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here: >> >> >> http://pastebin.com/gWrrS4zw >> >> Am not sure what is causing it. > > > > > From: mohammed shafeeque > > Subject: Re: [Freeswitch-users] Oneway audio issues in freeswitch > Date: February 11, 2016 at 7:28:45 PM GMT+3 > To: FreeSWITCH Users Help > > Reply-To: FreeSWITCH Users Help > > > > Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x > > On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz > wrote: > As a rule of dumb, try turning on rport > Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" > a ?crit : > You need to look at the sip signaling to see what's going on > > On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque > wrote: > Hello All > > We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. > > Here is scenario: > Grandstream call any extensions (one way audio) > Any extension call Grandstream ( Audio works just fine) > > We have tried multiple softphones and the result is same. > > Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. > > Any help or hint will be much appreciated. > > Thank you, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > ?talo Rossi > italo at freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/9e216e07/attachment-0001.html From ssinyagin at gmail.com Thu Feb 11 22:00:25 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 11 Feb 2016 20:00:25 +0100 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: hi Emrah and all, it's the first time I actually searched for it, but there are hosting offers with anycast IP routing. It means, you have multiple servers in various locations, and they share the same service IP address. The clients connect to the nearest server, which is determined by standard BGP routing. You are still limited to a single global hosting provider, but you benefit from its redundant network and geographical distribution. In case of anycast addressing, incoming connections will be served easily. But the outgoing connections are rather tricky: you will need to bring the outbound call to the physical server where the user has registered, and initiate the connection from its anycast address. So, you can share and replicate the registration database, but you need to send the outbound call to the server which accepted the registration. I guess you should be able to retrieve this information from the registration database. This needs to be looked in details. Google for anycast server hosting, and there are at least 3 providers offering virtual hosts, and OVH is offering physical hosts as well. I guess there are more providers with similar offerings. Without anycast, you would need to use redundant registrars sharing the same service IP address -- for example, Digitalocean offers such service within any single datacenter. Having multiple registrars with different IP addresses is also possible, but then you depend on the way how each particular SIP client handles multiple IP addresses after resolving the domain name. Some of them may get stuck to a single address, even if it's not responding. cheers, stanislav On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: > Hi list, > I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks. > > I am trying to achieve the following: > - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries. > - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense. > - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up. > > I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions. > > I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. > If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues. > > I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. > > Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless. > > Best, > E > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vipkilla at gmail.com Thu Feb 11 22:27:27 2016 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 11 Feb 2016 14:27:27 -0500 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: What about DNS SRV? On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin wrote: > hi Emrah and all, > > it's the first time I actually searched for it, but there are hosting > offers with anycast IP routing. It means, you have multiple servers in > various locations, and they share the same service IP address. The > clients connect to the nearest server, which is determined by standard > BGP routing. You are still limited to a single global hosting > provider, but you benefit from its redundant network and geographical > distribution. > > In case of anycast addressing, incoming connections will be served > easily. But the outgoing connections are rather tricky: you will need > to bring the outbound call to the physical server where the user has > registered, and initiate the connection from its anycast address. So, > you can share and replicate the registration database, but you need to > send the outbound call to the server which accepted the registration. > I guess you should be able to retrieve this information from the > registration database. This needs to be looked in details. > > Google for anycast server hosting, and there are at least 3 providers > offering virtual hosts, and OVH is offering physical hosts as well. I > guess there are more providers with similar offerings. > > > Without anycast, you would need to use redundant registrars sharing > the same service IP address -- for example, Digitalocean offers such > service within any single datacenter. > > Having multiple registrars with different IP addresses is also > possible, but then you depend on the way how each particular SIP > client handles multiple IP addresses after resolving the domain name. > Some of them may get stuck to a single address, even if it's not > responding. > > > cheers, > stanislav > > On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: > > Hi list, > > I?m writing to gather your thoughts and suggestions on how to have a > high availability FS setup on different networks. > > > > I am trying to achieve the following: > > - Load balance FreeSWITCH instances on 2 or more servers, possibly in > different countries. > > - Shared user directory and dialplan, but I?m not sure if shared > registrations would make sense. > > - If a server goes down, the phone should register on the alternative > servers. Obviously we can?t keep calls up. > > > > I?m obviously not the first one out there doing this. I?m trying to > learn from those who?ve come up with reliable solutions. > > > > I?ve tried sharing a registration table among multiple FS instances. But > it was a beginners mistake. Even with the right path to reach the client, > only the invites sent from the server used by the phone would be processed. > > If my phone registers on server A, then server A shares the info with > server B, server B knows how to contact the phone but it won?t be able to. > Supposedly because of NAT issues. > > > > I am aiming for fully independent FS instances that can back each other > up and be used independently. I am guessing this would require some sort of > SBC or external registrar server with a Kamailio or Repro. > > > > Anyway just trying to spark the conversation around this subject and > hopefully we can come up with a formula that can help many with their FS > deployments. My provider?s network just went all down in IPv4 and HA behind > the same provider proved to be useless. > > > > Best, > > E > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/8ff8fd96/attachment.html From s.safarov at gmail.com Thu Feb 11 23:16:45 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 11 Feb 2016 23:16:45 +0300 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: I use A record updated to point active FS servers in geo-distributed cluster. I DNS server i can recommend use Amazon Route53 On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa wrote: > What about DNS SRV? > > On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin > wrote: > >> hi Emrah and all, >> >> it's the first time I actually searched for it, but there are hosting >> offers with anycast IP routing. It means, you have multiple servers in >> various locations, and they share the same service IP address. The >> clients connect to the nearest server, which is determined by standard >> BGP routing. You are still limited to a single global hosting >> provider, but you benefit from its redundant network and geographical >> distribution. >> >> In case of anycast addressing, incoming connections will be served >> easily. But the outgoing connections are rather tricky: you will need >> to bring the outbound call to the physical server where the user has >> registered, and initiate the connection from its anycast address. So, >> you can share and replicate the registration database, but you need to >> send the outbound call to the server which accepted the registration. >> I guess you should be able to retrieve this information from the >> registration database. This needs to be looked in details. >> >> Google for anycast server hosting, and there are at least 3 providers >> offering virtual hosts, and OVH is offering physical hosts as well. I >> guess there are more providers with similar offerings. >> >> >> Without anycast, you would need to use redundant registrars sharing >> the same service IP address -- for example, Digitalocean offers such >> service within any single datacenter. >> >> Having multiple registrars with different IP addresses is also >> possible, but then you depend on the way how each particular SIP >> client handles multiple IP addresses after resolving the domain name. >> Some of them may get stuck to a single address, even if it's not >> responding. >> >> >> cheers, >> stanislav >> >> On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: >> > Hi list, >> > I?m writing to gather your thoughts and suggestions on how to have a >> high availability FS setup on different networks. >> > >> > I am trying to achieve the following: >> > - Load balance FreeSWITCH instances on 2 or more servers, possibly in >> different countries. >> > - Shared user directory and dialplan, but I?m not sure if shared >> registrations would make sense. >> > - If a server goes down, the phone should register on the alternative >> servers. Obviously we can?t keep calls up. >> > >> > I?m obviously not the first one out there doing this. I?m trying to >> learn from those who?ve come up with reliable solutions. >> > >> > I?ve tried sharing a registration table among multiple FS instances. >> But it was a beginners mistake. Even with the right path to reach the >> client, only the invites sent from the server used by the phone would be >> processed. >> > If my phone registers on server A, then server A shares the info with >> server B, server B knows how to contact the phone but it won?t be able to. >> Supposedly because of NAT issues. >> > >> > I am aiming for fully independent FS instances that can back each other >> up and be used independently. I am guessing this would require some sort of >> SBC or external registrar server with a Kamailio or Repro. >> > >> > Anyway just trying to spark the conversation around this subject and >> hopefully we can come up with a formula that can help many with their FS >> deployments. My provider?s network just went all down in IPv4 and HA behind >> the same provider proved to be useless. >> > >> > Best, >> > E >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/a8eef837/attachment-0001.html From vagarwal at vertical.com Fri Feb 12 01:00:57 2016 From: vagarwal at vertical.com (Varsha Agarwal) Date: Thu, 11 Feb 2016 22:00:57 +0000 Subject: [Freeswitch-users] Freeswitch in high available clustered environment Message-ID: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com> Hi All, Is there a good documentation on how to setup Freeswitch in a clustered environment with a redundant node as well? I am looking through Wiki but there no one good article I found that has it all. Thanks, Varsha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/a67ded35/attachment.html From steveayre at gmail.com Fri Feb 12 01:41:33 2016 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 11 Feb 2016 22:41:33 +0000 Subject: [Freeswitch-users] Two Users Registered ..But calls only going one way In-Reply-To: References: Message-ID: See the contents of the Contact header in A's REGISTER messages. They tell FreeSWITCH where to send the call to. NAT can confuse matters when picking the correct value though. It needs to be the external IP & port of the NAT router that the internal IP/port of the SIP messages are mapped to. Sometimes the phone will put the internal details instead which aren't routable externally. Sometimes it can detect it correctly (eg via STUN). If it can't some routers will contain a SIP ALG that will rewrite the header for a phone sending the internal ip/port to the external ip/port, but sometimes this can cause more problems than it solves if it doesn't do this correctly and it can't modify the packet if you're using TLS. On top of that that internal to external port mapping will expire on that NAT router if you don't re-REGISTER frequently enough so that could stop the INVITE getting through even if you're sending to the correct place. If you're having issues like that getting the SIP packets through then it's likely If you can't fix it on the phone/router then you can also look at the NDLB (no device left behind) options. For example there's one that'll use the address the REGISTER is received from instead of the Contact header. This differs from how SIP is supposed to work but works in most cases (usually a phone will ask you to call it directly not via a proxy or route you elsewhere). On 11 February 2016 at 09:46, David Wafula wrote: > Hi all, > I have two users who registered in the same domain: user A and B. > A can call B just fine. When B tries to call A, there is silence (no > ringback)..then after sometime the call goes into voice mail. A never > receives the call. Please see the call trace here: > > > http://pastebin.com/gWrrS4zw > > Am not sure what is causing it. > > Regards > > -- > David W > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/754a4692/attachment.html From blake at cogents.io Fri Feb 12 01:45:34 2016 From: blake at cogents.io (Blake Priddy) Date: Thu, 11 Feb 2016 16:45:34 -0600 Subject: [Freeswitch-users] Dev meeting Message-ID: https://www.gofundme.com/freeswitch It's not too late!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/376d82c4/attachment.html From luis.daniel.lucio at gmail.com Fri Feb 12 04:59:20 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 11 Feb 2016 20:59:20 -0500 Subject: [Freeswitch-users] Freeswitch in high available clustered environment In-Reply-To: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com> References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com> Message-ID: http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html Enjoy, contact me offline of you need more Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" a ?crit : > Hi All, > > > > Is there a good documentation on how to setup Freeswitch in a clustered > environment with a redundant node as well? I am looking through Wiki but > there no one good article I found that has it all. > > > > Thanks, > > Varsha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/551438ec/attachment.html From luis.daniel.lucio at gmail.com Fri Feb 12 05:40:09 2016 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 11 Feb 2016 21:40:09 -0500 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: Read this https://okay.com.mx/en/entrepreneurs/balancing-clustering-and-high-availability-with-fusionpbx.html It is just what you're looking for. Take the ideas and modify it to your needs Le 11 f?vr. 2016 3:17 PM, "Sergey Safarov" a ?crit : > I use A record updated to point active FS servers in geo-distributed > cluster. > I DNS server i can recommend use Amazon Route53 > > On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa wrote: > >> What about DNS SRV? >> >> On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin >> wrote: >> >>> hi Emrah and all, >>> >>> it's the first time I actually searched for it, but there are hosting >>> offers with anycast IP routing. It means, you have multiple servers in >>> various locations, and they share the same service IP address. The >>> clients connect to the nearest server, which is determined by standard >>> BGP routing. You are still limited to a single global hosting >>> provider, but you benefit from its redundant network and geographical >>> distribution. >>> >>> In case of anycast addressing, incoming connections will be served >>> easily. But the outgoing connections are rather tricky: you will need >>> to bring the outbound call to the physical server where the user has >>> registered, and initiate the connection from its anycast address. So, >>> you can share and replicate the registration database, but you need to >>> send the outbound call to the server which accepted the registration. >>> I guess you should be able to retrieve this information from the >>> registration database. This needs to be looked in details. >>> >>> Google for anycast server hosting, and there are at least 3 providers >>> offering virtual hosts, and OVH is offering physical hosts as well. I >>> guess there are more providers with similar offerings. >>> >>> >>> Without anycast, you would need to use redundant registrars sharing >>> the same service IP address -- for example, Digitalocean offers such >>> service within any single datacenter. >>> >>> Having multiple registrars with different IP addresses is also >>> possible, but then you depend on the way how each particular SIP >>> client handles multiple IP addresses after resolving the domain name. >>> Some of them may get stuck to a single address, even if it's not >>> responding. >>> >>> >>> cheers, >>> stanislav >>> >>> On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: >>> > Hi list, >>> > I?m writing to gather your thoughts and suggestions on how to have a >>> high availability FS setup on different networks. >>> > >>> > I am trying to achieve the following: >>> > - Load balance FreeSWITCH instances on 2 or more servers, possibly in >>> different countries. >>> > - Shared user directory and dialplan, but I?m not sure if shared >>> registrations would make sense. >>> > - If a server goes down, the phone should register on the alternative >>> servers. Obviously we can?t keep calls up. >>> > >>> > I?m obviously not the first one out there doing this. I?m trying to >>> learn from those who?ve come up with reliable solutions. >>> > >>> > I?ve tried sharing a registration table among multiple FS instances. >>> But it was a beginners mistake. Even with the right path to reach the >>> client, only the invites sent from the server used by the phone would be >>> processed. >>> > If my phone registers on server A, then server A shares the info with >>> server B, server B knows how to contact the phone but it won?t be able to. >>> Supposedly because of NAT issues. >>> > >>> > I am aiming for fully independent FS instances that can back each >>> other up and be used independently. I am guessing this would require some >>> sort of SBC or external registrar server with a Kamailio or Repro. >>> > >>> > Anyway just trying to spark the conversation around this subject and >>> hopefully we can come up with a formula that can help many with their FS >>> deployments. My provider?s network just went all down in IPv4 and HA behind >>> the same provider proved to be useless. >>> > >>> > Best, >>> > E >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160211/1815a2cd/attachment-0001.html From vagarwal at vertical.com Fri Feb 12 05:59:25 2016 From: vagarwal at vertical.com (Varsha Agarwal) Date: Fri, 12 Feb 2016 02:59:25 +0000 Subject: [Freeswitch-users] Freeswitch in high available clustered environment In-Reply-To: References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com>, Message-ID: Thanks I will review it. On Feb 11, 2016, at 6:01 PM, Luis Daniel Lucio Quiroz > wrote: http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html Enjoy, contact me offline of you need more Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" > a ?crit : Hi All, Is there a good documentation on how to setup Freeswitch in a clustered environment with a redundant node as well? I am looking through Wiki but there no one good article I found that has it all. Thanks, Varsha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/90c02721/attachment.html From max at nysolutions.com Fri Feb 12 06:13:54 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 12 Feb 2016 03:13:54 +0000 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: Luis, This mailing list is a Freeswitch users list, this list is for users to help each other with freeswitch, not a place to advertise your services, you may want to look at the freeswitch biz list http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: Thursday, February 11, 2016 9:40 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] High availability on different networks Read this https://okay.com.mx/en/entrepreneurs/balancing-clustering-and-high-availability-with-fusionpbx.html It is just what you're looking for. Take the ideas and modify it to your needs Le 11 f?vr. 2016 3:17 PM, "Sergey Safarov" > a ?crit : I use A record updated to point active FS servers in geo-distributed cluster. I DNS server i can recommend use Amazon Route53 On Thu, Feb 11, 2016 at 10:27 PM, Vik Killa > wrote: What about DNS SRV? On Thu, Feb 11, 2016 at 2:00 PM, Stanislav Sinyagin > wrote: hi Emrah and all, it's the first time I actually searched for it, but there are hosting offers with anycast IP routing. It means, you have multiple servers in various locations, and they share the same service IP address. The clients connect to the nearest server, which is determined by standard BGP routing. You are still limited to a single global hosting provider, but you benefit from its redundant network and geographical distribution. In case of anycast addressing, incoming connections will be served easily. But the outgoing connections are rather tricky: you will need to bring the outbound call to the physical server where the user has registered, and initiate the connection from its anycast address. So, you can share and replicate the registration database, but you need to send the outbound call to the server which accepted the registration. I guess you should be able to retrieve this information from the registration database. This needs to be looked in details. Google for anycast server hosting, and there are at least 3 providers offering virtual hosts, and OVH is offering physical hosts as well. I guess there are more providers with similar offerings. Without anycast, you would need to use redundant registrars sharing the same service IP address -- for example, Digitalocean offers such service within any single datacenter. Having multiple registrars with different IP addresses is also possible, but then you depend on the way how each particular SIP client handles multiple IP addresses after resolving the domain name. Some of them may get stuck to a single address, even if it's not responding. cheers, stanislav On Thu, Feb 11, 2016 at 5:53 PM, Emrah > wrote: > Hi list, > I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks. > > I am trying to achieve the following: > - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries. > - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense. > - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up. > > I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions. > > I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. > If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues. > > I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. > > Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless. > > Best, > E > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/67d55c1f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/67d55c1f/attachment-0001.jpg From bilaln018 at gmail.com Fri Feb 12 09:06:38 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 12 Feb 2016 11:06:38 +0500 Subject: [Freeswitch-users] [SDP Negotiation Error][Calling through JsSIP] In-Reply-To: References: Message-ID: Hi All, For the community archive i have added following in sofia internal profile and its started working. Regards Abbasi On Thu, Feb 11, 2016 at 7:54 PM, Bilal Abbasi wrote: > Thanks brain for reply but what i need to do, how can i get rid of this > warning. > > many thanks > > abbasi > > > On Thursday, February 11, 2016, Brian West wrote: > > NO candidate ACL defined, Defaulting to wan.auto >> sofia/internal/1001 at 192.241.213.201:7000 no suitable candidates found. >> >> On Thu, Feb 11, 2016 at 3:19 AM, Bilal Abbasi >> wrote: >> >>> Hi all, >>> >>> Currently i am facing an CODEC NEGOTIATION ERROR while calling through >>> JsSIP Caller 1001 Callie 1000, >>> Please view the logs, >>> https://pastebin.freeswitch.org/24550 >>> >>> I have enabled PCMA and PCMU in my var.xml, >>> >>> Regards >>> Abbasi >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/498e1c6e/attachment.html From ssinyagin at gmail.com Fri Feb 12 10:40:53 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 12 Feb 2016 08:40:53 +0100 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: there is an issue with anycast routing though: when you bring up a new server, some running sessions will be dropped, because their IP packets would be routed to a different host. So, it needs a careful design. Maybe place only the SIP proxy on an anycast address, and run the calls from unique local addresses. Multiple DNS SRV records with different priorities are also possible, but you can't direct the users to the nearest location within the same domain. Also a bunch of SIP clients needs to be tested and you need to answer the questions, like: -- what is the timeout if the primary server is unavailable? -- if the primary host goes down during the call, how soon can the client re-dial? -- what happens if the primary server comes up again? On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin wrote: > hi Emrah and all, > > it's the first time I actually searched for it, but there are hosting > offers with anycast IP routing. It means, you have multiple servers in > various locations, and they share the same service IP address. The > clients connect to the nearest server, which is determined by standard > BGP routing. You are still limited to a single global hosting > provider, but you benefit from its redundant network and geographical > distribution. > > In case of anycast addressing, incoming connections will be served > easily. But the outgoing connections are rather tricky: you will need > to bring the outbound call to the physical server where the user has > registered, and initiate the connection from its anycast address. So, > you can share and replicate the registration database, but you need to > send the outbound call to the server which accepted the registration. > I guess you should be able to retrieve this information from the > registration database. This needs to be looked in details. > > Google for anycast server hosting, and there are at least 3 providers > offering virtual hosts, and OVH is offering physical hosts as well. I > guess there are more providers with similar offerings. > > > Without anycast, you would need to use redundant registrars sharing > the same service IP address -- for example, Digitalocean offers such > service within any single datacenter. > > Having multiple registrars with different IP addresses is also > possible, but then you depend on the way how each particular SIP > client handles multiple IP addresses after resolving the domain name. > Some of them may get stuck to a single address, even if it's not > responding. > > > cheers, > stanislav > > On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: >> Hi list, >> I?m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks. >> >> I am trying to achieve the following: >> - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries. >> - Shared user directory and dialplan, but I?m not sure if shared registrations would make sense. >> - If a server goes down, the phone should register on the alternative servers. Obviously we can?t keep calls up. >> >> I?m obviously not the first one out there doing this. I?m trying to learn from those who?ve come up with reliable solutions. >> >> I?ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. >> If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won?t be able to. Supposedly because of NAT issues. >> >> I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. >> >> Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider?s network just went all down in IPv4 and HA behind the same provider proved to be useless. >> >> Best, >> E >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From andrew at cassidywebservices.co.uk Fri Feb 12 12:09:32 2016 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 12 Feb 2016 09:09:32 +0000 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: It's not instant, but I've used OVH failover IP's to do that sort of thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that does the OVH API call to move the IP address. Sadly it's not that quick, takes about 2 minutes. On 12 February 2016 at 07:40, Stanislav Sinyagin wrote: > there is an issue with anycast routing though: when you bring up a new > server, some running sessions will be dropped, because their IP > packets would be routed to a different host. So, it needs a careful > design. Maybe place only the SIP proxy on an anycast address, and run > the calls from unique local addresses. > > > Multiple DNS SRV records with different priorities are also possible, > but you can't direct the users to the nearest location within the same > domain. Also a bunch of SIP clients needs to be tested and you need to > answer the questions, like: > > -- what is the timeout if the primary server is unavailable? > -- if the primary host goes down during the call, how soon can the > client re-dial? > -- what happens if the primary server comes up again? > > > > > > > On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin > wrote: > > hi Emrah and all, > > > > it's the first time I actually searched for it, but there are hosting > > offers with anycast IP routing. It means, you have multiple servers in > > various locations, and they share the same service IP address. The > > clients connect to the nearest server, which is determined by standard > > BGP routing. You are still limited to a single global hosting > > provider, but you benefit from its redundant network and geographical > > distribution. > > > > In case of anycast addressing, incoming connections will be served > > easily. But the outgoing connections are rather tricky: you will need > > to bring the outbound call to the physical server where the user has > > registered, and initiate the connection from its anycast address. So, > > you can share and replicate the registration database, but you need to > > send the outbound call to the server which accepted the registration. > > I guess you should be able to retrieve this information from the > > registration database. This needs to be looked in details. > > > > Google for anycast server hosting, and there are at least 3 providers > > offering virtual hosts, and OVH is offering physical hosts as well. I > > guess there are more providers with similar offerings. > > > > > > Without anycast, you would need to use redundant registrars sharing > > the same service IP address -- for example, Digitalocean offers such > > service within any single datacenter. > > > > Having multiple registrars with different IP addresses is also > > possible, but then you depend on the way how each particular SIP > > client handles multiple IP addresses after resolving the domain name. > > Some of them may get stuck to a single address, even if it's not > > responding. > > > > > > cheers, > > stanislav > > > > On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: > >> Hi list, > >> I?m writing to gather your thoughts and suggestions on how to have a > high availability FS setup on different networks. > >> > >> I am trying to achieve the following: > >> - Load balance FreeSWITCH instances on 2 or more servers, possibly in > different countries. > >> - Shared user directory and dialplan, but I?m not sure if shared > registrations would make sense. > >> - If a server goes down, the phone should register on the alternative > servers. Obviously we can?t keep calls up. > >> > >> I?m obviously not the first one out there doing this. I?m trying to > learn from those who?ve come up with reliable solutions. > >> > >> I?ve tried sharing a registration table among multiple FS instances. > But it was a beginners mistake. Even with the right path to reach the > client, only the invites sent from the server used by the phone would be > processed. > >> If my phone registers on server A, then server A shares the info with > server B, server B knows how to contact the phone but it won?t be able to. > Supposedly because of NAT issues. > >> > >> I am aiming for fully independent FS instances that can back each other > up and be used independently. I am guessing this would require some sort of > SBC or external registrar server with a Kamailio or Repro. > >> > >> Anyway just trying to spark the conversation around this subject and > hopefully we can come up with a formula that can help many with their FS > deployments. My provider?s network just went all down in IPv4 and HA behind > the same provider proved to be useless. > >> > >> Best, > >> E > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/53bcd9a3/attachment-0001.html From aqsyounas at gmail.com Fri Feb 12 12:14:36 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 12 Feb 2016 14:14:36 +0500 Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch. Message-ID: Hi, I am bridging Invite containing both RPID and PID to some destination. Like Currently, I see freeswitch can add only one header PID or RPID but not both with sip_cid_type variable. How can i Add both? Remote-Party-ID: ;privacy=off;screen=yes P-Asserted-Identity: Any suggestion is much appreciate. Best Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/8d608964/attachment.html From aqsyounas at gmail.com Fri Feb 12 13:50:48 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 12 Feb 2016 15:50:48 +0500 Subject: [Freeswitch-users] fs_path adding route header Message-ID: Hi, I am using fs_path to proxy Invite to specified proxy. But I see fs_path adding route header in Invite. As B2BUA freeswitch must be generating new call instead of adding route header. Is this intended behavior? Or something wrong with my configuration Best Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/7f60c91f/attachment.html From ssinyagin at gmail.com Fri Feb 12 14:49:17 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 12 Feb 2016 12:49:17 +0100 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: as Sergey has proposed, there could be a DNS service which monitors the availability of your VoIP servers and changes the DNS entries if a server goes down. The TTL for individual SRV records could be set to few seconds. But that means again that all users are using the same server, so it's not really a distributed model as Emrah challenged in the original mail. Probably this new project will help in building a distributed cluster, but it needs a detailed study: https://ipfs.io/ On Fri, Feb 12, 2016 at 10:09 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > It's not instant, but I've used OVH failover IP's to do that sort of > thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that > does the OVH API call to move the IP address. > > Sadly it's not that quick, takes about 2 minutes. > > On 12 February 2016 at 07:40, Stanislav Sinyagin > wrote: > >> there is an issue with anycast routing though: when you bring up a new >> server, some running sessions will be dropped, because their IP >> packets would be routed to a different host. So, it needs a careful >> design. Maybe place only the SIP proxy on an anycast address, and run >> the calls from unique local addresses. >> >> >> Multiple DNS SRV records with different priorities are also possible, >> but you can't direct the users to the nearest location within the same >> domain. Also a bunch of SIP clients needs to be tested and you need to >> answer the questions, like: >> >> -- what is the timeout if the primary server is unavailable? >> -- if the primary host goes down during the call, how soon can the >> client re-dial? >> -- what happens if the primary server comes up again? >> >> >> >> >> >> >> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin >> wrote: >> > hi Emrah and all, >> > >> > it's the first time I actually searched for it, but there are hosting >> > offers with anycast IP routing. It means, you have multiple servers in >> > various locations, and they share the same service IP address. The >> > clients connect to the nearest server, which is determined by standard >> > BGP routing. You are still limited to a single global hosting >> > provider, but you benefit from its redundant network and geographical >> > distribution. >> > >> > In case of anycast addressing, incoming connections will be served >> > easily. But the outgoing connections are rather tricky: you will need >> > to bring the outbound call to the physical server where the user has >> > registered, and initiate the connection from its anycast address. So, >> > you can share and replicate the registration database, but you need to >> > send the outbound call to the server which accepted the registration. >> > I guess you should be able to retrieve this information from the >> > registration database. This needs to be looked in details. >> > >> > Google for anycast server hosting, and there are at least 3 providers >> > offering virtual hosts, and OVH is offering physical hosts as well. I >> > guess there are more providers with similar offerings. >> > >> > >> > Without anycast, you would need to use redundant registrars sharing >> > the same service IP address -- for example, Digitalocean offers such >> > service within any single datacenter. >> > >> > Having multiple registrars with different IP addresses is also >> > possible, but then you depend on the way how each particular SIP >> > client handles multiple IP addresses after resolving the domain name. >> > Some of them may get stuck to a single address, even if it's not >> > responding. >> > >> > >> > cheers, >> > stanislav >> > >> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: >> >> Hi list, >> >> I?m writing to gather your thoughts and suggestions on how to have a >> high availability FS setup on different networks. >> >> >> >> I am trying to achieve the following: >> >> - Load balance FreeSWITCH instances on 2 or more servers, possibly in >> different countries. >> >> - Shared user directory and dialplan, but I?m not sure if shared >> registrations would make sense. >> >> - If a server goes down, the phone should register on the alternative >> servers. Obviously we can?t keep calls up. >> >> >> >> I?m obviously not the first one out there doing this. I?m trying to >> learn from those who?ve come up with reliable solutions. >> >> >> >> I?ve tried sharing a registration table among multiple FS instances. >> But it was a beginners mistake. Even with the right path to reach the >> client, only the invites sent from the server used by the phone would be >> processed. >> >> If my phone registers on server A, then server A shares the info with >> server B, server B knows how to contact the phone but it won?t be able to. >> Supposedly because of NAT issues. >> >> >> >> I am aiming for fully independent FS instances that can back each >> other up and be used independently. I am guessing this would require some >> sort of SBC or external registrar server with a Kamailio or Repro. >> >> >> >> Anyway just trying to spark the conversation around this subject and >> hopefully we can come up with a formula that can help many with their FS >> deployments. My provider?s network just went all down in IPv4 and HA behind >> the same provider proved to be useless. >> >> >> >> Best, >> >> E >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/a1919cf9/attachment-0001.html From blake at cogents.io Fri Feb 12 16:44:44 2016 From: blake at cogents.io (Blake Priddy) Date: Fri, 12 Feb 2016 07:44:44 -0600 Subject: [Freeswitch-users] Over the half way mark!!! :D In-Reply-To: References: Message-ID: They are over the half way mark! Thanks y'all. Keep it coming!! :) https://www.gofundme.com/freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/9d0e50f7/attachment.html From mike at jerris.com Fri Feb 12 19:03:33 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 Feb 2016 11:03:33 -0500 Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch. In-Reply-To: References: Message-ID: I don't think we have any way exposed to do this right now, but I wouldn't be opposed to a patch that added this On Friday, February 12, 2016, Aqs Younas wrote: > Hi, > > I am bridging Invite containing both RPID and PID to some destination. Like > > data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/> > > Currently, I see freeswitch can add only one header PID or RPID but not > both with sip_cid_type variable. > > How can i Add both? > > Remote-Party-ID: > >;privacy=off;screen=yes > P-Asserted-Identity: > ;user=phone> > > Any suggestion is much appreciate. > > Best Regards. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/9fda7afc/attachment.html From s.safarov at gmail.com Fri Feb 12 19:14:44 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 12 Feb 2016 16:14:44 +0000 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: As cluster solution I use kazoo cluster with kamailio - after 6 mouth of usage I can say it perfect solution. Also updates DNS record can be implemented via custom corosync resource agent script. Write this script is task of two or tree days. This solution is not require external servers. On Fri, Feb 12, 2016, 14:50 Stanislav Sinyagin wrote: > as Sergey has proposed, there could be a DNS service which monitors the > availability of your VoIP servers and changes the DNS entries if a server > goes down. The TTL for individual SRV records could be set to few seconds. > > But that means again that all users are using the same server, so it's not > really a distributed model as Emrah challenged in the original mail. > > Probably this new project will help in building a distributed cluster, but > it needs a detailed study: https://ipfs.io/ > > > > > > > On Fri, Feb 12, 2016 at 10:09 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> It's not instant, but I've used OVH failover IP's to do that sort of >> thing. Heartbeat over ipsec tunnel, use pacemaker with a custom script that >> does the OVH API call to move the IP address. >> >> Sadly it's not that quick, takes about 2 minutes. >> >> On 12 February 2016 at 07:40, Stanislav Sinyagin >> wrote: >> >>> there is an issue with anycast routing though: when you bring up a new >>> server, some running sessions will be dropped, because their IP >>> packets would be routed to a different host. So, it needs a careful >>> design. Maybe place only the SIP proxy on an anycast address, and run >>> the calls from unique local addresses. >>> >>> >>> Multiple DNS SRV records with different priorities are also possible, >>> but you can't direct the users to the nearest location within the same >>> domain. Also a bunch of SIP clients needs to be tested and you need to >>> answer the questions, like: >>> >>> -- what is the timeout if the primary server is unavailable? >>> -- if the primary host goes down during the call, how soon can the >>> client re-dial? >>> -- what happens if the primary server comes up again? >>> >>> >>> >>> >>> >>> >>> On Thu, Feb 11, 2016 at 8:00 PM, Stanislav Sinyagin >>> wrote: >>> > hi Emrah and all, >>> > >>> > it's the first time I actually searched for it, but there are hosting >>> > offers with anycast IP routing. It means, you have multiple servers in >>> > various locations, and they share the same service IP address. The >>> > clients connect to the nearest server, which is determined by standard >>> > BGP routing. You are still limited to a single global hosting >>> > provider, but you benefit from its redundant network and geographical >>> > distribution. >>> > >>> > In case of anycast addressing, incoming connections will be served >>> > easily. But the outgoing connections are rather tricky: you will need >>> > to bring the outbound call to the physical server where the user has >>> > registered, and initiate the connection from its anycast address. So, >>> > you can share and replicate the registration database, but you need to >>> > send the outbound call to the server which accepted the registration. >>> > I guess you should be able to retrieve this information from the >>> > registration database. This needs to be looked in details. >>> > >>> > Google for anycast server hosting, and there are at least 3 providers >>> > offering virtual hosts, and OVH is offering physical hosts as well. I >>> > guess there are more providers with similar offerings. >>> > >>> > >>> > Without anycast, you would need to use redundant registrars sharing >>> > the same service IP address -- for example, Digitalocean offers such >>> > service within any single datacenter. >>> > >>> > Having multiple registrars with different IP addresses is also >>> > possible, but then you depend on the way how each particular SIP >>> > client handles multiple IP addresses after resolving the domain name. >>> > Some of them may get stuck to a single address, even if it's not >>> > responding. >>> > >>> > >>> > cheers, >>> > stanislav >>> > >>> > On Thu, Feb 11, 2016 at 5:53 PM, Emrah wrote: >>> >> Hi list, >>> >> I?m writing to gather your thoughts and suggestions on how to have a >>> high availability FS setup on different networks. >>> >> >>> >> I am trying to achieve the following: >>> >> - Load balance FreeSWITCH instances on 2 or more servers, possibly >>> in different countries. >>> >> - Shared user directory and dialplan, but I?m not sure if shared >>> registrations would make sense. >>> >> - If a server goes down, the phone should register on the alternative >>> servers. Obviously we can?t keep calls up. >>> >> >>> >> I?m obviously not the first one out there doing this. I?m trying to >>> learn from those who?ve come up with reliable solutions. >>> >> >>> >> I?ve tried sharing a registration table among multiple FS instances. >>> But it was a beginners mistake. Even with the right path to reach the >>> client, only the invites sent from the server used by the phone would be >>> processed. >>> >> If my phone registers on server A, then server A shares the info with >>> server B, server B knows how to contact the phone but it won?t be able to. >>> Supposedly because of NAT issues. >>> >> >>> >> I am aiming for fully independent FS instances that can back each >>> other up and be used independently. I am guessing this would require some >>> sort of SBC or external registrar server with a Kamailio or Repro. >>> >> >>> >> Anyway just trying to spark the conversation around this subject and >>> hopefully we can come up with a formula that can help many with their FS >>> deployments. My provider?s network just went all down in IPv4 and HA behind >>> the same provider proved to be useless. >>> >> >>> >> Best, >>> >> E >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://confluence.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/e553f08c/attachment-0001.html From gmaruzz at gmail.com Fri Feb 12 19:25:00 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 12 Feb 2016 17:25:00 +0100 Subject: [Freeswitch-users] FUND THE PARTY! @ FreeSWITCH Annual Core Devs Meeting Message-ID: How much we all made in 2015 thanks to FreeSWITCH? How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? Shovel your HARD EARNED MONEY to our HARD PARTYING and HARD CODING CORE DEVS ! BUY THEM LUNCH AND REFRESHMENTS ! HERE! ==> https://www.gofundme.com/freeswitch If not now, when? Because! -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/7d010b43/attachment.html From aqsyounas at gmail.com Fri Feb 12 19:35:26 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 12 Feb 2016 21:35:26 +0500 Subject: [Freeswitch-users] How to add both rpid and pid in freeswitch. In-Reply-To: References: Message-ID: Thanks Micheal. I am able to achieve above by setting paid as header and adding rpid as sip_cid_type in bridge command. On 12 February 2016 at 21:03, Michael Jerris wrote: > I don't think we have any way exposed to do this right now, but I wouldn't > be opposed to a patch that added this > > > On Friday, February 12, 2016, Aqs Younas wrote: > >> Hi, >> >> I am bridging Invite containing both RPID and PID to some destination. >> Like >> >> > data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/> >> >> Currently, I see freeswitch can add only one header PID or RPID but not >> both with sip_cid_type variable. >> >> How can i Add both? >> >> Remote-Party-ID: ;privacy=off;screen=yes >> P-Asserted-Identity: >> >> Any suggestion is much appreciate. >> >> Best Regards. >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/1e2051ff/attachment.html From aqsyounas at gmail.com Fri Feb 12 21:24:26 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 12 Feb 2016 23:24:26 +0500 Subject: [Freeswitch-users] fs_path adding route header In-Reply-To: References: Message-ID: freeswitch is adding route header of value fs_path. Which is causing problem in 200ok and ack. Is there any way to proxy request without making freeswitch add route header. Or any other way to make proxy request to destination other than RURI without changing RURI. Best Regards. On 12 February 2016 at 15:50, Aqs Younas wrote: > Hi, > > I am using fs_path to proxy Invite to specified proxy. But I see fs_path > adding route header in Invite. As B2BUA freeswitch must be generating new > call instead of adding route header. > > Is this intended behavior? Or something wrong with my configuration > > Best Regards. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/a856035c/attachment.html From lists at telefaks.de Fri Feb 12 22:56:17 2016 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 12 Feb 2016 20:56:17 +0100 Subject: [Freeswitch-users] FUND THE PARTY! @ FreeSWITCH Annual Core Devs Meeting In-Reply-To: References: Message-ID: <56BE38E1.6070303@telefaks.de> Just transferred another thank you via Paypal. Enjoy your meal! Best regards Peter On 02/12/16 17:25, Giovanni Maruzzelli wrote: > How much we all made in 2015 thanks to FreeSWITCH? > > How much we made thanks to core devs (Anthony, Mike, Ken, Brian, > William, etc) answering our difficult questions? > > Shovel your HARD EARNED MONEY to our HARD PARTYING and HARD CODING > CORE DEVS ! > > BUY THEM LUNCH AND REFRESHMENTS ! > > HERE! ==> https://www.gofundme.com/freeswitch > > If not now, when? > > Because! > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/aa5b4c5e/attachment.html From brian at freeswitch.org Sat Feb 13 02:50:18 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Feb 2016 17:50:18 -0600 Subject: [Freeswitch-users] Freeswitch in high available clustered environment In-Reply-To: References: <04AB0A185A23864CA9255FD38901F65D023374576B@SCEX1.vertical.com> Message-ID: We're actually working on a doc for this, it should be out soon. On Thu, Feb 11, 2016 at 7:59 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html > > Enjoy, contact me offline of you need more > Le 11 f?vr. 2016 5:01 PM, "Varsha Agarwal" a > ?crit : > >> Hi All, >> >> >> >> Is there a good documentation on how to setup Freeswitch in a clustered >> environment with a redundant node as well? I am looking through Wiki but >> there no one good article I found that has it all. >> >> >> >> Thanks, >> >> Varsha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/fcbe0988/attachment-0001.html From 568691 at gmail.com Sat Feb 13 04:11:54 2016 From: 568691 at gmail.com (Alexandru Covalschi) Date: Sat, 13 Feb 2016 03:11:54 +0200 Subject: [Freeswitch-users] Get actual duration of the call after hangup Message-ID: Hello folks! I have a problem - I need to get actual duration of the call - I mean from ANSWER to HANGUP, without the time user waited being parked on the bridge. The thing is I need to write that info into a custom table which belongs to a custom database where I have leg_a uuid. How can I achieve that in case of: 1. Outoing origination via bridge() 2. Connecting two customers which are already inside a freeswitch via bridge() I've read about mod_cdr_csv but it afaik doesn't allow to connect to a custom db. Would mod_odbc_cdr allow me to do that? And if yes - can you tell me "chan-var-name"? And don't be afraid - such slow method is not for billing :) Just to provide customer more detailed calls history. Thanks! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160213/72044a97/attachment.html From brian at freeswitch.org Sat Feb 13 05:00:15 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Feb 2016 20:00:15 -0600 Subject: [Freeswitch-users] Get actual duration of the call after hangup In-Reply-To: References: Message-ID: Parse the xml cdr it will give you more than you want probably. On Friday, February 12, 2016, Alexandru Covalschi <568691 at gmail.com> wrote: > Hello folks! > > I have a problem - I need to get actual duration of the call - I mean from > ANSWER to HANGUP, without the time user waited being parked on the bridge. > The thing is I need to write that info into a custom table which belongs to > a custom database where I have leg_a uuid. > How can I achieve that in case of: > 1. Outoing origination via bridge() > 2. Connecting two customers which are already inside a freeswitch via > bridge() > I've read about mod_cdr_csv but it afaik doesn't allow to connect to a > custom db. > Would mod_odbc_cdr allow me to do that? And if yes - can you tell me > "chan-var-name"? > > And don't be afraid - such slow method is not for billing :) Just to > provide customer more detailed calls history. > > Thanks! > -- > Alexandru Covalschi > ABRISS-Solutions > VoIP engineer and system administrator > phone: +37367398493 > web: http://abs-telecom.com/ > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/2044bca7/attachment.html From rutu.patel at inextrix.com Sat Feb 13 07:43:12 2016 From: rutu.patel at inextrix.com (Rutu Patel) Date: Sat, 13 Feb 2016 10:13:12 +0530 Subject: [Freeswitch-users] Call recording audio too fast Message-ID: Hi All, I am using record_session application to record the calls but the audio speed has too fast in recording files. Below is the code I am using for call recording: Can anyone please help me to solve the issue with fast audio speed in recording files ? Also is it possible to decrease the size of .wav audio file ? -- Thanks, Rutu Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160213/91b81efd/attachment.html From brian at freeswitch.org Sat Feb 13 07:55:30 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Feb 2016 22:55:30 -0600 Subject: [Freeswitch-users] Call recording audio too fast In-Reply-To: References: Message-ID: Oh great swami nanda, what revision are you running? We would need a few more details to know for sure, have you tried the latest master? On Friday, February 12, 2016, Rutu Patel wrote: > Hi All, > > I am using record_session application to record the calls but the audio > speed has too fast in recording files. > > Below is the code I am using for call recording: > > > > > > Can anyone please help me to solve the issue with fast audio speed in > recording files ? > Also is it possible to decrease the size of .wav audio file ? > -- > Thanks, > Rutu Patel > > > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/6b5c3580/attachment.html From rutu.patel at inextrix.com Sat Feb 13 08:10:00 2016 From: rutu.patel at inextrix.com (Rutu Patel) Date: Sat, 13 Feb 2016 10:40:00 +0530 Subject: [Freeswitch-users] Call recording audio too fast In-Reply-To: References: Message-ID: Hi Brian, I am using freeswitch-1.6.5 on Debian-8.2. What further information I should share ? -- Thanks, Rutu Patel On Sat, Feb 13, 2016 at 10:25 AM, Brian West wrote: > Oh great swami nanda, what revision are you running? We would need a few > more details to know for sure, have you tried the latest master? > > > On Friday, February 12, 2016, Rutu Patel wrote: > >> Hi All, >> >> I am using record_session application to record the calls but the audio >> speed has too fast in recording files. >> >> Below is the code I am using for call recording: >> >> >> >> >> >> Can anyone please help me to solve the issue with fast audio speed in >> recording files ? >> Also is it possible to decrease the size of .wav audio file ? >> -- >> Thanks, >> Rutu Patel >> >> >> > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160213/b8c484ed/attachment-0001.html From brian at freeswitch.org Sat Feb 13 08:15:52 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Feb 2016 23:15:52 -0600 Subject: [Freeswitch-users] Call recording audio too fast In-Reply-To: References: Message-ID: Can you try 1.6.6 then post logs if a call? On Friday, February 12, 2016, Rutu Patel wrote: > Hi Brian, > > I am using freeswitch-1.6.5 on Debian-8.2. > What further information I should share ? > > -- > Thanks, > Rutu Patel > > > On Sat, Feb 13, 2016 at 10:25 AM, Brian West > wrote: > >> Oh great swami nanda, what revision are you running? We would need a few >> more details to know for sure, have you tried the latest master? >> >> >> On Friday, February 12, 2016, Rutu Patel > > wrote: >> >>> Hi All, >>> >>> I am using record_session application to record the calls but the audio >>> speed has too fast in recording files. >>> >>> Below is the code I am using for call recording: >>> >>> >>> >>> >>> >>> Can anyone please help me to solve the issue with fast audio speed in >>> recording files ? >>> Also is it possible to decrease the size of .wav audio file ? >>> -- >>> Thanks, >>> Rutu Patel >>> >>> >>> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160212/14f5a5b2/attachment.html From jprangi at didforsale.com Sat Feb 13 23:23:38 2016 From: jprangi at didforsale.com (Jai Rangi) Date: Sat, 13 Feb 2016 12:23:38 -0800 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: Found the problem, We had this enabled in profiles. auto-jitterbuffer-msec value=60 Commenting it seems to have fixed the issue. Not sure why this would cause problem only from one type of phones. Any hint. Thank you, *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Sat, Feb 13, 2016 at 9:45 AM, Jai Rangi wrote: > Hello Ken, > Thank for look in this. Attached are debug logs. SIP Traces were not > molested, except the public IPs were changed. As of writing of this email, > the issue is isolated to 1.6.x. > Not sure if anyone else has tested this on latest version. But easy to > reproduce. Just download grandstream Wave, available to IOS and Andriod and > try to call any extension directly. Curious to see if any one can come with > different result. > > *Jai Rangi* > Cebod Technologies LLC dba DIDforSale/Cebod Telecom > O 949-471-0102 | C 949-419-7634 | F > 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | > www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | > > > > > > > On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice wrote: > >> without logs of a call doing this at debug level with a complete >> unmolested sip trace in line its a little hard to speculate whats going on >> here >> >> On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque > > wrote: >> >>> Surprised that no one else experienced this problem. Can anyone give any >>> hint. Really Dont want to move back to 1.4.x >>> >>> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < >>> luis.daniel.lucio at gmail.com> wrote: >>> >>>> As a rule of dumb, try turning on rport >>>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" a >>>> ?crit : >>>> >>>>> You need to look at the sip signaling to see what's going on >>>>> >>>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque < >>>>> shafeeq.v at gmail.com> wrote: >>>>> >>>>>> Hello All >>>>>> >>>>>> We are getting one way audio issues with some softphones and >>>>>> grandstream phones behind nat registerd to our freeswitch server. >>>>>> >>>>>> Here is scenario: >>>>>> Grandstream call any extensions (one way audio) >>>>>> Any extension call Grandstream ( Audio works just fine) >>>>>> >>>>>> We have tried multiple softphones and the result is same. >>>>>> >>>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having >>>>>> DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue >>>>>> started with an upgrade to freeswitch. >>>>>> >>>>>> Any help or hint will be much appreciated. >>>>>> >>>>>> Thank you, >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ?talo Rossi >>>>> italo at freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160213/7dce293f/attachment-0001.html From jprangi at didforsale.com Sat Feb 13 20:45:43 2016 From: jprangi at didforsale.com (Jai Rangi) Date: Sat, 13 Feb 2016 09:45:43 -0800 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: Hello Ken, Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result. *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice wrote: > without logs of a call doing this at debug level with a complete > unmolested sip trace in line its a little hard to speculate whats going on > here > > On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque > wrote: > >> Surprised that no one else experienced this problem. Can anyone give any >> hint. Really Dont want to move back to 1.4.x >> >> On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < >> luis.daniel.lucio at gmail.com> wrote: >> >>> As a rule of dumb, try turning on rport >>> Le 10 f?vr. 2016 8:55 PM, "?talo Rossi" a ?crit : >>> >>>> You need to look at the sip signaling to see what's going on >>>> >>>> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque < >>>> shafeeq.v at gmail.com> wrote: >>>> >>>>> Hello All >>>>> >>>>> We are getting one way audio issues with some softphones and >>>>> grandstream phones behind nat registerd to our freeswitch server. >>>>> >>>>> Here is scenario: >>>>> Grandstream call any extensions (one way audio) >>>>> Any extension call Grandstream ( Audio works just fine) >>>>> >>>>> We have tried multiple softphones and the result is same. >>>>> >>>>> Everything was working fine with 1.4.18 and 1.6.2. We were having >>>>> DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue >>>>> started with an upgrade to freeswitch. >>>>> >>>>> Any help or hint will be much appreciated. >>>>> >>>>> Thank you, >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> ?talo Rossi >>>> italo at freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160213/05c78853/attachment-0001.html -------------- next part -------------- 2016-02-13 09:22:50.747900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv 2016-02-13 09:22:50.747900 [NOTICE] mod_logfile.c:213 New log started. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1276 at domain.example.com [a17a5d15-d97e-4c70-b476-bd0ff4ac2e09] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_NEW a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] sofia.c:9248 sofia/internal/1276 at domain.example.com receiving invite from 68.5.94.63:12113 version: 1.6.6 64bit a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:413 Looking for zrtp-hash a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2016-02-13 09:23:10.157920 [DEBUG] sofia.c:9415 IP 68.5.94.63 Rejected by acl "domains". Falling back to Digest auth. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.157920 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/1276 at domain.example.com) State NEW 2016-02-13 09:23:10.157920 [DEBUG] sofia.c:2147 detaching session a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.387894 [DEBUG] sofia.c:2255 Re-attaching to session a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [DEBUG] sofia.c:9248 sofia/internal/1276 at domain.example.com receiving invite from 68.5.94.63:12113 version: 1.6.6 64bit a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:413 Looking for zrtp-hash a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.397894 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2016-02-13 09:23:10.397894 [DEBUG] sofia.c:9415 IP 68.5.94.63 Rejected by acl "domains". Falling back to Digest auth. 2016-02-13 09:23:10.397894 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:23:10.407800 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [received][100] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:6770 Remote SDP: a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=1276 21012 1 IN IP4 192.168.1.168 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=- a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 192.168.1.168 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 8002 RTP/AVP 0 18 8 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:18 G729/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:8 PCMA/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] sofia.c:7125 (sofia/internal/1276 at domain.example.com) State Change CS_NEW -> CS_INIT a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_INIT a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at domain.example.com) State INIT a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:88 sofia/internal/1276 at domain.example.com SOFIA INIT a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1276 at domain.example.com Standard INIT a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1276 at domain.example.com) State Change CS_INIT -> CS_ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at domain.example.com) State INIT going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:2247 (sofia/internal/1276 at domain.example.com) Callstate Change DOWN -> RINGING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at domain.example.com SOFIA ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1276 at domain.example.com Standard ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [INFO] mod_dialplan_xml.c:637 Processing Kunal Mittal <1276>->142 in context public a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->limit_exceeded] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [limit_exceeded] destination_number(142) =~ /^limit_exceeded$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_testopensip] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [from_testopensip] network_addr(68.5.94.63) =~ /^209\.216\.15\.18$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_productionopensip] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [from_productionopensip] network_addr(68.5.94.63) =~ /^209\.216\.2\.222$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [public->from_registeredusers] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [from_registeredusers] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(dialed_domain=${sip_to_host}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(domain_name=${sip_to_host}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(company_name=${sip_to_host}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(caller-id-in-from=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(sip_cid_type=none) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(Public IP is ${bind_server_ip} Domain is ${domain_name} Dialed domain is ${dialed_domain}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action lua(didhandle.lua ${destination_number}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1276 at domain.example.com) State Change CS_ROUTING -> CS_EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_sofia.c:196 sofia/internal/1276 at domain.example.com SOFIA EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1276 at domain.example.com Standard EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(dialed_domain=domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [dialed_domain]=[domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(domain_name=domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [domain_name]=[domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(company_name=domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [company_name]=[domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(caller-id-in-from=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(sip_cid_type=none) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com log(Public IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] mod_dptools.c:1692 IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com lua(didhandle.lua 142) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] 2016-02-13 09:23:10.407800 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(domain_name=domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(skip_cdr_causes=LOSE_RACE) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [INFO] switch_cpp.cpp:1284 RPID is dialing number is 1276 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(accountcode=17) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.407800 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [INFO] switch_cpp.cpp:1284 String length is less that 7 should be extension a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(process_cdr=a_only) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [process_cdr]=[a_only] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_ivr.c:2085 (sofia/internal/1276 at domain.example.com) State Change CS_EXECUTE -> CS_ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_ivr.c:2090 sofia/internal/1276 at domain.example.com receive message [TRANSFER] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [NOTICE] switch_ivr.c:2092 Transfer sofia/internal/1276 at domain.example.com to XML[142 at default] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_cpp.cpp:898 transfer result: 0 2016-02-13 09:23:10.417792 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_cpp.cpp:1103 sofia/internal/1276 at domain.example.com destroy/unlink session from object a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at domain.example.com SOFIA ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1276 at domain.example.com Standard ROUTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [INFO] mod_dialplan_xml.c:637 Processing Kunal Mittal <1276>->142 in context default a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->longdistance] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [longdistance] destination_number(142) =~ /^\d{10,15}$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unloop] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->freeswitch_public_conf_via_sip] continue=false 2016-02-13 09:23:10.417792 [ERR] switch_regex.c:104 COMPILE ERROR: 1 [nothing to repeat][^*9(888|8888|1616|3232)$] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(142) =~ /^*9(888|8888|1616|3232)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->tod_example] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Date/TimeMatch (FAIL) [tod_example] break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->holiday_example] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Date/TimeMatch (FAIL) [holiday_example] break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->global-intercept] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global-intercept] destination_number(142) =~ /^\*886$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->group-intercept] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [group-intercept] destination_number(142) =~ /^\*8$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->intercept-ext] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [intercept-ext] destination_number(142) =~ /^\*\*(\d+)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->redial] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [redial] destination_number(142) =~ /^\*(redial|870)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->global] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [global] ${switch_r_sdp}(v=0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=1276 21012 1 IN IP4 192.168.1.168 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=- a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 192.168.1.168 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 8002 RTP/AVP 0 18 8 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:18 G729/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:8 PCMA/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [global] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->snom-demo-2] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [snom-demo-2] destination_number(142) =~ /^9001$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->snom-demo-1] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [snom-demo-1] destination_number(142) =~ /^9000$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->eavesdrop] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [eavesdrop] destination_number(142) =~ /^\*88(\d{3,5})$|^\*0(.*)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->eavesdrop] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [eavesdrop] destination_number(142) =~ /^\*779$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call_return] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call_return] destination_number(142) =~ /^\*69$|^\*869$|^lcr$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->del-number] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [del-number] destination_number(142) =~ /^\*60$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->add-number] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [add-number] destination_number(142) =~ /^\*61$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->check-number] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [check-number] destination_number(142) =~ /^\*62$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->enable-time-condition] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [enable-time-condition] destination_number(142) =~ /^\*63$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->disable-time-condition] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [disable-time-condition] destination_number(142) =~ /^\*64$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->del-group] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [del-group] destination_number(142) =~ /^\*80$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->add-group] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [add-group] destination_number(142) =~ /^\*81$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call-group-simo] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call-group-simo] destination_number(142) =~ /^\*82(\d{2})$|^82(\d{2})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->call-group-order] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [call-group-order] destination_number(142) =~ /^\*83(\d{2})$|^83(\d{2})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->extension-intercom] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [extension-intercom] destination_number(142) =~ /^\*84(\d{3,5})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->user_recodring_enabled] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [user_recodring_enabled] ${recording}(off) =~ /^(on|off)$/ break=on-true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->user_callscreening] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [user_callscreening] caller_id_number(1276) =~ /^\d{10,15}$/ break=on-true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [user_callscreening] ${callscreening}(off) =~ /^(on|off)$/ break=on-true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Check IVR-based CF] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [Check IVR-based CF] destination_number(142) =~ /^(\d+)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(dialed_number=142) INLINE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(dialed_number=142) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [dialed_number]=[142] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(cf_target=${db(select/${domain_name}-CF/142)}) INLINE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(cf_target=) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [cf_target]=[UNDEF] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${recording}(off) =~ /^on$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{3,5}$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{10,15}$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF from my station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF from my station] destination_number(142) =~ /^\*72$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF cancel from my station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF cancel from my station] destination_number(142) =~ /^\*73$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF from any station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF from any station] destination_number(142) =~ /^\*76$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->CF cancel from any station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [CF cancel from any station] destination_number(142) =~ /^\*77$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->NACF from my station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [NACF from my station] destination_number(142) =~ /^\*74$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->NACF cancel from my station] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [NACF cancel from my station] destination_number(142) =~ /^\*75$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Add_Member_To_Queue] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Add_Member_To_Queue] destination_number(142) =~ /^\*51$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Delete_Member_From_Queue] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Delete_Member_From_Queue] destination_number(142) =~ /^\*52$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Pickup_Call_From_Queue] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Pickup_Call_From_Queue] destination_number(142) =~ /^\*53$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->dial-by-name] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [dial-by-name] destination_number(142) =~ /^\*40$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->conference_withpin] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [conference_withpin] destination_number(142) =~ /^\*3560$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Voicemail_check] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Voicemail_check] caller_id_number(1276) =~ /^(142)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->check_availability] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [check_availability] destination_number(142) =~ /^hkjhk(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->is_check_availability] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [is_check_availability] available_to_call() =~ /true/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->osbridge] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [osbridge] destination_number(142) =~ /(^\d{3,5})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(proxy_media=false) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(continue_on_fail=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(dialed_extension=142) INLINE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(dialed_extension=142) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[142] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(domain_name=${domain_name}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(accountcode=${user_data(${dialed_extension}@${domain_name} var accountcode)}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(available_to_call=true) INLINE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(available_to_call=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [available_to_call]=[true] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action log(Extensions ${dialed_extension} is ${available_to_call}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(nacf_target=${db(select/${domain_name}-NACF/142)}) INLINE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(nacf_target=) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [nacf_target]=[UNDEF] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${recording}(off) =~ /^on$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [osbridge] ${available_to_call}(true) =~ /^true$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(1 b s execute_extension::dx XML features) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/${domain_name}/${strftime(%Y-%m-%d-%H-%M-%S)}.${uuid}.wav) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(3 b s execute_extension::cf XML features) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(4 b s execute_extension::attented_xfer XML default) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(6 b s execute_extension::blind_xfer_to XML default) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bind_meta_app(9 b s execute_extension::call_park XML default) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action export(sip_contact_user=${sip_from_user}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(ringback=${us-ring}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(transfer_ringback=local_stream://moh) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(call_timeout=35) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action set(hangup_after_bridge=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action bridge(user/${dialed_extension}@${domain_name}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action sleep(1000) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{3,7}$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{10,15}$/ break=never a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Absolute Condition [osbridge] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action answer() a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action sleep(500) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action voicemail(default ${domain_name} ${dialed_extension}) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension_VM] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension_VM] destination_number(142) =~ /^vm-(\d{3,5})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->attented_xfer_to_vm] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [attented_xfer_to_vm] destination_number(142) =~ /^attented_xfer_to_vm$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->attented_xfer] continue=true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [attented_xfer] destination_number(142) =~ /^attented_xfer$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->blind_xfer_to] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [blind_xfer_to] destination_number(142) =~ /^blind_xfer_to$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension] destination_number(142) =~ /^disable-(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Local_Extension_Skinny] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Local_Extension_Skinny] destination_number(142) =~ /^(11[01][0-9])$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park-in] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park-in] destination_number(142) =~ /^call_park$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park-out] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park-out] destination_number(142) =~ /^\*(85\d\d)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->vmain] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [vmain] destination_number(142) =~ /^vmain$|^\*98$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->sip_uri] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [sip_uri] destination_number(142) =~ /^sip:(.*)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->local_uri] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [local_uri] destination_number(142) =~ /^(local_.*)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->nb_conferences] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [nb_conferences] destination_number(142) =~ /^\*(30\d{2,4})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->wb_conferences] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [wb_conferences] destination_number(142) =~ /^\*(31\d{2})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->uwb_conferences] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [uwb_conferences] destination_number(142) =~ /^\*(32\d{2})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->cdquality_conferences] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [cdquality_conferences] destination_number(142) =~ /^\*(33\d{2})$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss_intercom] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss_intercom] destination_number(142) =~ /^0911$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss_intercom] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss_intercom] destination_number(142) =~ /^0912$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->mad_boss] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [mad_boss] destination_number(142) =~ /^0913$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->queupark] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [queupark] destination_number(142) =~ /^\*(5902)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->queueunpark] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [queueunpark] destination_number(142) =~ /^\*(5903)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /^\*5900$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /^\*(5901)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->valet_park] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [valet_park] destination_number(142) =~ /^(6000)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->valet_park] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [valet_park] destination_number(142) =~ /^(60\d[1-9])$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /park\+(\d+)/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /^parking$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->park] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [park] destination_number(142) =~ /callpark/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->unpark] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [unpark] destination_number(142) =~ /pickup/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->wait] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [wait] destination_number(142) =~ /^wait$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->fax_receive] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [fax_receive] destination_number(142) =~ /^fax.*$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->fax_transmit] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [fax_transmit] destination_number(142) =~ /^9179$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_180] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_180] destination_number(142) =~ /^9180$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_183_uk_ring] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_183_uk_ring] destination_number(142) =~ /^\*9181$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_183_music_ring] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_183_music_ring] destination_number(142) =~ /^\*9182$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_post_answer_uk_ring] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(142) =~ /^\*9183$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->ringback_post_answer_music] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [ringback_post_answer_music] destination_number(142) =~ /^\*9184$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->show_info] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [show_info] destination_number(142) =~ /^\*9192$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->video_record] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [video_record] destination_number(142) =~ /^\*9193$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->video_playback] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [video_playback] destination_number(142) =~ /^\*9194$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->delay_echo] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [delay_echo] destination_number(142) =~ /^\*9195$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->echo] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [echo] destination_number(142) =~ /^\*9196$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->milliwatt] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [milliwatt] destination_number(142) =~ /^\*9197$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->tone_stream] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [tone_stream] destination_number(142) =~ /^\*9198$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->zrtp_enrollement] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [zrtp_enrollement] destination_number(142) =~ /^\*9787$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->hold_music] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [hold_music] destination_number(142) =~ /^\*9664$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->laugh break] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [laugh break] destination_number(142) =~ /^\*9386$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->101] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [101] destination_number(142) =~ /^101$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->pizza_demo] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [pizza_demo] destination_number(142) =~ /^(pizza|74992)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Time] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Time] destination_number(142) =~ /^9170$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Date] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Date] destination_number(142) =~ /^9171$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->Talking Clock Date and Time] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [Talking Clock Date and Time] destination_number(142) =~ /^9172$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->local.example.com] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->domestic.example.com] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->international.example.com] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->conference] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [conference] destination_number(142) =~ /^112233$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->didforsale_ivr] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [didforsale_ivr] destination_number(142) =~ /5566/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->domian_ivr] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (FAIL) [domian_ivr] destination_number(142) =~ /ivr(1949.*)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com parsing [default->enum] continue=false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Regex (PASS) [enum] destination_number(142) =~ /^(.*)$/ break=on-false a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 Dialplan: sofia/internal/1276 at domain.example.com Action transfer(142 enum) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1276 at domain.example.com) State Change CS_ROUTING -> CS_EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at domain.example.com) State ROUTING going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1276 at domain.example.com) State EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_sofia.c:196 sofia/internal/1276 at domain.example.com SOFIA EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1276 at domain.example.com Standard EXECUTE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-spymap/1276/a17a5d15-d97e-4c70-b476-bd0ff4ac2e09) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-last_dial/1276/142) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com hash(insert/domain.example.com-last_dial/global/a17a5d15-d97e-4c70-b476-bd0ff4ac2e09) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(RFC2822_DATE=Sat, 13 Feb 2016 09:23:10 -0800) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(proxy_media=false) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [proxy_media]=[false] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(continue_on_fail=true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [continue_on_fail]=[true] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com export(domain_name=domain.example.com) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.417792 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] 2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:23:10.427795 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com set(accountcode=17) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1276 at domain.example.com [accountcode]=[17] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com log(Extensions 142 is true) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] mod_dptools.c:1692 142 is true a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(1 b s execute_extension::dx XML features) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-23-10.a17a5d15-d97e-4c70-b476-bd0ff4ac2e09.wav) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2781 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-23-10.a17a5d15-d97e-4c70-b476-bd0ff4ac2e09.wav a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [ALERT] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 E