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 EXECUTE sofia/internal/1276 at domain.example.com bind_meta_app(3 b s execute_extension::cf 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: *3 execute_extension::cf 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(4 b s execute_extension::attented_xfer XML default) 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: *4 execute_extension::attented_xfer XML default 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(5 b s execute_extension::attented_xfer_to_vm XML default) 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: *5 execute_extension::attented_xfer_to_vm XML default 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(6 b s execute_extension::blind_xfer_to XML default) 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: *6 execute_extension::blind_xfer_to XML default 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(9 b s execute_extension::call_park XML default) 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: *9 execute_extension::call_park XML default 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 export(sip_contact_user=1276) 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] switch_channel.c:1293 EXPORT (export_vars) [sip_contact_user]=[1276] 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 set(ringback=%(2000,4000,440,480)) 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 [ringback]=[%(2000,4000,440,480)] 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 set(transfer_ringback=local_stream://moh) 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 [transfer_ringback]=[local_stream://moh] 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 set(call_timeout=35) 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 [call_timeout]=[35] 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 set(hangup_after_bridge=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:1498 SET sofia/internal/1276 at domain.example.com [hangup_after_bridge]=[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 bridge(user/142 at domain.example.com) 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] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [caller-id-in-from]=[true] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_cid_type]=[none] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [accountcode]=[17] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [dialed_extension]=[142] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [available_to_call]=[true] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_contact_user]=[1276] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables 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 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [caller-id-in-from]=[true] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_cid_type]=[none] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [accountcode]=[17] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [dialed_extension]=[142] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [available_to_call]=[true] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:23:10 -0800] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [domain_name]=[domain.example.com] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_channel.c:1247 sofia/internal/1276 at domain.example.com EXPORTING[export_vars] [sip_contact_user]=[1276] to event a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.427795 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.427795 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/142 at 192.168.33.116:49383 [d9770cc8-1742-40cd-876f-3b5f50733449] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.427795 [DEBUG] mod_sofia.c:4776 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_NEW -> CS_INIT d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_INIT d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at 192.168.33.116:49383) State INIT d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] mod_sofia.c:88 sofia/internal/142 at 192.168.33.116:49383 SOFIA INIT d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia_glue.c:1228 sip:142 at 73.170.102.43:49383 Setting proxy route to sofia/internal/142 at 192.168.33.116:49383 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia_glue.c:1257 sofia/internal/142 at 192.168.33.116:49383 sending invite version: 1.6.6 64bit d9770cc8-1742-40cd-876f-3b5f50733449 Local SDP: d9770cc8-1742-40cd-876f-3b5f50733449 v=0 d9770cc8-1742-40cd-876f-3b5f50733449 o=FreeSWITCH 1455355946 1455355947 IN IP4 219.206.20.22 d9770cc8-1742-40cd-876f-3b5f50733449 s=FreeSWITCH d9770cc8-1742-40cd-876f-3b5f50733449 c=IN IP4 219.206.20.22 d9770cc8-1742-40cd-876f-3b5f50733449 t=0 0 d9770cc8-1742-40cd-876f-3b5f50733449 m=audio 28244 RTP/AVP 0 18 8 101 13 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:0 PCMU/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:18 G729/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:8 PCMA/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:101 telephone-event/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=fmtp:101 0-16 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:13 CN/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=ptime:20 d9770cc8-1742-40cd-876f-3b5f50733449 a=sendrecv d9770cc8-1742-40cd-876f-3b5f50733449 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:40 sofia/internal/142 at 192.168.33.116:49383 Standard INIT d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_INIT -> CS_ROUTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at 192.168.33.116:49383) State INIT going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_ROUTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [calling][0] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at 192.168.33.116:49383) State ROUTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] mod_sofia.c:141 sofia/internal/142 at 192.168.33.116:49383 SOFIA ROUTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_ROUTING -> CS_CONSUME_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at 192.168.33.116:49383) State ROUTING going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_CONSUME_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/142 at 192.168.33.116:49383) State CONSUME_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/142 at 192.168.33.116:49383) State CONSUME_MEDIA going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.437790 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! 2016-02-13 09:23:10.507868 [ALERT] sofia.c:1214 sofia/internal/142 at 192.168.33.116:49383 Same Callee ID "Outbound Call" <142> d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [proceeding][180] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/142 at 192.168.33.116:49383! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [DEBUG] switch_channel.c:3340 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change DOWN -> RINGING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] sofia.c:6862 sofia/internal/142 at 192.168.33.116:49383 receive message [RING_EVENT] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:10.507868 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_ivr_originate.c:1216 sofia/internal/1276 at domain.example.com receive message [PROGRESS] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [INFO] switch_ivr_originate.c:1216 Sending early media a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:413 Looking for zrtp-hash a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 96 at 8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1276 at domain.example.com PCMU/8000 20 ms 160 samples 64000 bits 1 channels a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_codec.c:111 sofia/internal/1276 at domain.example.com Original read codec set to PCMU:0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 96 at 8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:4485 sofia/internal/1276 at domain.example.com Set 2833 dtmf send payload to 96 recv payload to 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1276 at domain.example.com] 219.206.20.22 port 26876 -> 192.168.1.168 port 8002 codec: 0 ms: 20 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6332 sofia/internal/1276 at domain.example.com Set 2833 dtmf send payload to 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6339 sofia/internal/1276 at domain.example.com Set 2833 dtmf receive payload to 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_media.c:6362 sofia/internal/1276 at domain.example.com Set rtp dtmf delay to 60 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1276 at domain.example.com! a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_channel.c:3468 (sofia/internal/1276 at domain.example.com) Callstate Change RINGING -> EARLY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [ALERT] sofia_media.c:92 sofia/internal/1276 at domain.example.com receive message [PROGRESS_EVENT] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] mod_sofia.c:2330 Ring SDP: a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=FreeSWITCH 1455357314 1455357315 IN IP4 219.206.20.22 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=FreeSWITCH a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 219.206.20.22 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 26876 RTP/AVP 0 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=fmtp:96 0-16 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=sendrecv a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_core_codec.c:221 sofia/internal/1276 at domain.example.com Push codec L16:100 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.517869 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [early][183] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [ALERT] switch_channel.c:3507 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.537870 [ALERT] switch_core_io.c:1135 sofia/internal/1276 at domain.example.com receive message [TRANSCODING_NECESSARY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:10.827958 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.1.168:8002 to 68.5.94.63:8002 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:13.517891 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57074 but expected: 57072 lost: 2 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:413 Looking for zrtp-hash d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.537897 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 2016-02-13 09:23:14.547896 [ALERT] sofia.c:1214 sofia/internal/142 at 192.168.33.116:49383 Same Callee ID "Outbound Call" <142> d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [completing][200] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6770 Remote SDP: d9770cc8-1742-40cd-876f-3b5f50733449 v=0 d9770cc8-1742-40cd-876f-3b5f50733449 o=142 8000 8000 IN IP4 192.168.33.116 d9770cc8-1742-40cd-876f-3b5f50733449 s=SIP Call d9770cc8-1742-40cd-876f-3b5f50733449 c=IN IP4 192.168.33.116 d9770cc8-1742-40cd-876f-3b5f50733449 t=0 0 d9770cc8-1742-40cd-876f-3b5f50733449 m=audio 12552 RTP/AVP 0 8 101 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:0 PCMU/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:8 PCMA/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=rtpmap:101 telephone-event/8000 d9770cc8-1742-40cd-876f-3b5f50733449 a=fmtp:101 0-15 d9770cc8-1742-40cd-876f-3b5f50733449 a=ptime:20 d9770cc8-1742-40cd-876f-3b5f50733449 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at 192.168.33.116:49383 entering state [ready][200] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:413 Looking for zrtp-hash d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/142 at 192.168.33.116:49383 PCMU/8000 20 ms 160 samples 64000 bits 1 channels d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_codec.c:111 sofia/internal/142 at 192.168.33.116:49383 Original read codec set to PCMU:0 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:4485 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf send payload to 101 recv payload to 101 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/142 at 192.168.33.116:49383] 219.206.20.22 port 28244 -> 192.168.33.116 port 12552 codec: 0 ms: 20 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6332 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf send payload to 101 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6339 sofia/internal/142 at 192.168.33.116:49383 Set 2833 dtmf receive payload to 101 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_core_media.c:6362 sofia/internal/142 at 192.168.33.116:49383 Set rtp dtmf delay to 60 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [NOTICE] sofia.c:7724 Channel [sofia/internal/142 at 192.168.33.116:49383] has been answered d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [DEBUG] switch_channel.c:3767 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change RINGING -> ACTIVE d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] sofia.c:7724 sofia/internal/142 at 192.168.33.116:49383 receive message [ANSWER_EVENT] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.547896 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_core_codec.c:246 sofia/internal/1276 at domain.example.com Restore previous codec PCMU:0. a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3550 sofia/internal/1276 at domain.example.com receive message [ANSWER] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/1276 at domain.example.com: a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 v=0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 o=FreeSWITCH 1455357314 1455357316 IN IP4 219.206.20.22 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 s=FreeSWITCH a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 c=IN IP4 219.206.20.22 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 t=0 0 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 m=audio 26876 RTP/AVP 0 96 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:0 PCMU/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=rtpmap:96 telephone-event/8000 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=fmtp:96 0-16 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=ptime:20 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a=sendrecv a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/1276 at domain.example.com] has been answered a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_channel.c:3767 (sofia/internal/1276 at domain.example.com) Callstate Change EARLY -> ACTIVE a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3550 sofia/internal/1276 at domain.example.com receive message [ANSWER_EVENT] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [completed][200] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/142 at 192.168.33.116:49383] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3944 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3941 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/142 at 192.168.33.116:49383] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3944 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_originate.c:3941 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1452 sofia/internal/142 at 192.168.33.116:49383 receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:590 sofia/internal/142 at 192.168.33.116:49383 session thread sleep state: CS_CONSUME_MEDIA! a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr.c:217 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1451 sofia/internal/1276 at domain.example.com receive message [AUDIO_SYNC] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1551 sofia/internal/142 at 192.168.33.116:49383 receive message [BRIDGE] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_media.c:9156 sofia/internal/142 at 192.168.33.116:49383 PAUSE Jitterbuffer a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [ALERT] switch_ivr_bridge.c:1559 sofia/internal/1276 at domain.example.com receive message [BRIDGE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.557902 [DEBUG] switch_core_media.c:9156 sofia/internal/1276 at domain.example.com PAUSE Jitterbuffer d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [ALERT] switch_core_state_machine.c:594 sofia/internal/142 at 192.168.33.116:49383 session thread wake state: CS_CONSUME_MEDIA! d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_EXCHANGE_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/142 at 192.168.33.116:49383) State EXCHANGE_MEDIA d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.557902 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.577825 [ALERT] switch_core_io.c:420 sofia/internal/1276 at domain.example.com receive message [TRANSCODING_NECESSARY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.657897 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at domain.example.com entering state [ready][200] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:14.667847 [ALERT] switch_ivr_bridge.c:283 sofia/internal/142 at 192.168.33.116:49383 receive message [DISPLAY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.677907 [ALERT] switch_ivr_bridge.c:283 sofia/internal/1276 at domain.example.com receive message [DISPLAY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:14.697815 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed. d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:15.367824 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.33.116:12552 to 73.170.102.43:12552 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:15.507826 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 17 but expected: 16 lost: 1 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:15.617825 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57095 but expected: 57093 lost: 2 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:16.437897 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57116 but expected: 57115 lost: 1 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:17.747957 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 129 but expected: 128 lost: 1 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:18.987891 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 191 but expected: 190 lost: 1 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:19.307915 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57160 but expected: 57155 lost: 5 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:20.797895 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57194 but expected: 57192 lost: 2 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:21.247947 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 304 but expected: 303 lost: 1 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:22.467916 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 365 but expected: 364 lost: 1 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:22.617927 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57247 but expected: 57245 lost: 2 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:23.477916 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 416 but expected: 415 lost: 1 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:23.897894 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57297 but expected: 57296 lost: 1 a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:24.757895 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at domain.example.com Got: audio seq 57315 but expected: 57313 lost: 2 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:25.747894 [ALERT] switch_rtp.c:1612 sofia/internal/142 at 192.168.33.116:49383 Got: audio seq 529 but expected: 528 lost: 1 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/+12127773456 at 66.62.60.228:5060 [022e5d6e-83a2-4397-84ba-f0f49aa6eef6] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_NEW 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:9248 sofia/internal/+12127773456 at 66.62.60.228:5060 receiving invite from 209.216.2.222:5660 version: 1.6.6 64bit 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:9360 IP 209.216.2.222 Approved by acl "domains[]". Access Granted. 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [received][100] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:6770 Remote SDP: 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 v=0 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 o=Sonus_UAC 28164 6556 IN IP4 66.62.60.228 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 s=SIP Media Capabilities 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 c=IN IP4 66.62.60.229 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 t=0 0 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 m=audio 11570 RTP/AVP 18 0 101 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:18 G729/8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:18 annexb=no 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:0 PCMU/8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:101 telephone-event/8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:101 0-15 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=maxptime:20 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] sofia.c:7125 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_NEW -> CS_INIT 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/+12127773456 at 66.62.60.228:5060) State NEW 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_INIT 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/+12127773456 at 66.62.60.228:5060) State INIT 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:88 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA INIT 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:40 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard INIT 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_INIT -> CS_ROUTING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/+12127773456 at 66.62.60.228:5060) State INIT going to sleep 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_ROUTING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:2247 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change DOWN -> RINGING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/+12127773456 at 66.62.60.228:5060) State ROUTING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:141 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA ROUTING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:166 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard ROUTING 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [INFO] mod_dialplan_xml.c:637 Processing FOOD BANK NYC <+12127773456>->19499300360 in context public 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->limit_exceeded] continue=false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (FAIL) [limit_exceeded] destination_number(19499300360) =~ /^limit_exceeded$/ break=on-false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->from_testopensip] continue=false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (FAIL) [from_testopensip] network_addr(209.216.2.222) =~ /^209\.216\.15\.18$/ break=on-false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 parsing [public->from_productionopensip] continue=false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Regex (PASS) [from_productionopensip] network_addr(209.216.2.222) =~ /^209\.216\.2\.222$/ break=on-false 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(dialed_domain=${sip_to_host}) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(domain_name=${sip_to_host}) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action set(company_name=${sip_to_host}) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action export(caller-id-in-from=true) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action export(sip_cid_type=none) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action lua(didhandle.lua ${destination_number}) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 Dialplan: sofia/internal/+12127773456 at 66.62.60.228:5060 Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain}) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/+12127773456 at 66.62.60.228:5060) State Change CS_ROUTING -> CS_EXECUTE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/+12127773456 at 66.62.60.228:5060) State ROUTING going to sleep 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/+12127773456 at 66.62.60.228:5060) Running State Change CS_EXECUTE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/+12127773456 at 66.62.60.228:5060) State EXECUTE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_sofia.c:196 sofia/internal/+12127773456 at 66.62.60.228:5060 SOFIA EXECUTE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_core_state_machine.c:258 sofia/internal/+12127773456 at 66.62.60.228:5060 Standard EXECUTE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(dialed_domain=209.216.15.70) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [dialed_domain]=[209.216.15.70] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(domain_name=209.216.15.70) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [domain_name]=[209.216.15.70] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(company_name=209.216.15.70) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [company_name]=[209.216.15.70] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(caller-id-in-from=true) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(sip_cid_type=none) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 lua(didhandle.lua 19499300360) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 2016-02-13 09:23:25.807894 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(domain_name=209.216.15.70) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[209.216.15.70] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(skip_cdr_causes=LOSE_RACE) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [INFO] switch_cpp.cpp:1284 RPID is ;privacy=off dialing number is +12127773456 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 0 1/10) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_limit.c:126 incr called: +12127773456_0 max:1, interval:10 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_hash.c:198 Usage for +12127773456_0 is now 1/1 for the last 10 seconds 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 0 3/60) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] switch_limit.c:126 incr called: +12127773456_0 max:3, interval:60 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(effective_caller_id_name=FOOD BANK NYC) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [effective_caller_id_name]=[FOOD BANK NYC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.807894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 2016-02-13 09:23:25.817894 [NOTICE] switch_cpp.cpp:1356 Product List Query SQL: select cebod_card.hold_music,cebod_card.status, cp.iduser, cp.product, cp.producttype as producttype, cp.activated, cd.domain, lower(cp.destinationtype) as destinationtype, cp.destination, cp.timecondition, cp.timeid, cp.timedestinationtype, cp.timedestination, cd.pbxip as pbxip, recording, whisper, whisper_file, postcall_id, substring(grp_description,1,15) as grpdescription from cebod_products as cp, domain as cd left join cebod_card on cebod_card.id=iduser where cp.iduser=cd.iduser and substr(cp.product,length(cp.product)-9,10) = substr('19499300360',length('19499300360')-9,10) 2016-02-13 09:23:25.817894 [NOTICE] switch_cpp.cpp:1356 Time Availability : yes 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 set(api_hangup_hook=lua postcall.lua 1 domain.example.com) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_dptools.c:1498 SET sofia/internal/+12127773456 at 66.62.60.228:5060 [api_hangup_hook]=[lua postcall.lua 1 domain.example.com] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [INFO] switch_cpp.cpp:1284 Check id called id is blocked or not , notblockedSQL is select * from cebod_callblock where domain=domain.example.com and ( upper(phonenumber)=upper(substr(12127773456,1,length(phonenumber))) or upper(phonenumber)=upper(substr(FOOD BANK NYC,1,length(phonenumber))) ) limit 1 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [INFO] switch_cpp.cpp:1284 Check id called id+12127773456 is blocked or not , notblocked 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(calltype=origination) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [calltype]=[origination] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash +12127773456 17 1/10) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_limit.c:126 incr called: +12127773456_17 max:1, interval:10 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_hash.c:198 Usage for +12127773456_17 is now 1/1 for the last 10 seconds 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(domain_name=domain.example.com) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(accountcode=17) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 export(setinboundcalllimit=setlimitdomain.example.com 17 15) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [setinboundcalllimit]=[setlimitdomain.example.com 17 15] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 limit(hash domain.example.com 17 15) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] switch_limit.c:126 incr called: domain.example.com_17 max:15, interval:0 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [DEBUG] mod_hash.c:196 Usage for domain.example.com_17 is now 1/15 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:25.817894 [ALERT] switch_core_session.c:2796 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC_COMPLETE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.327895 [NOTICE] sofia.c:952 Hangup sofia/internal/1276 at domain.example.com [CS_EXECUTE] [NORMAL_CLEARING] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/1276 at domain.example.com ending bridge by request from write function d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/142 at 192.168.33.116:49383] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/142 at 192.168.33.116:49383 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/142 at 192.168.33.116:49383) State EXCHANGE_MEDIA going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_HANGUP d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/142 at 192.168.33.116:49383) Callstate Change ACTIVE -> HANGUP d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at 192.168.33.116:49383) State HANGUP d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:425 sofia/internal/142 at 192.168.33.116:49383 Overriding SIP cause 480 with 200 from the other leg d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:431 Channel sofia/internal/142 at 192.168.33.116:49383 hanging up, cause: NORMAL_CLEARING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/142 at 192.168.33.116:49383 d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:60 sofia/internal/142 at 192.168.33.116:49383 Standard HANGUP, cause: NORMAL_CLEARING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at 192.168.33.116:49383) State HANGUP going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_HANGUP -> CS_REPORTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_REPORTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at 192.168.33.116:49383) State REPORTING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:104 sofia/internal/142 at 192.168.33.116:49383 Standard REPORTING, cause: NORMAL_CLEARING d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at 192.168.33.116:49383) State REPORTING going to sleep d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/142 at 192.168.33.116:49383) State Change CS_REPORTING -> CS_DESTROY d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.327895 [DEBUG] switch_core_session.c:1646 Session 1857 (sofia/internal/142 at 192.168.33.116:49383) Locked, Waiting on external entities a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:705 sofia/internal/1276 at domain.example.com ending bridge by request from read function a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1276 at domain.example.com] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [ALERT] switch_ivr_bridge.c:1692 sofia/internal/1276 at domain.example.com receive message [UNBRIDGE] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/1276 at domain.example.com skip receive message [UNBRIDGE] (channel is hungup already) d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [NOTICE] switch_core_session.c:1664 Session 1857 (sofia/internal/142 at 192.168.33.116:49383) Ended d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/142 at 192.168.33.116:49383 [CS_DESTROY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [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 09:23:26.337895 [DEBUG] switch_core_session.c:2796 sofia/internal/1276 at domain.example.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [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 09:23:26.337895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_HANGUP d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/142 at 192.168.33.116:49383) Running State Change CS_DESTROY d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at 192.168.33.116:49383) State DESTROY d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] mod_sofia.c:341 sofia/internal/142 at 192.168.33.116:49383 SOFIA DESTROY d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:111 sofia/internal/142 at 192.168.33.116:49383 Standard DESTROY d9770cc8-1742-40cd-876f-3b5f50733449 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at 192.168.33.116:49383) State DESTROY going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1276 at domain.example.com) Callstate Change ACTIVE -> HANGUP a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at domain.example.com) State HANGUP a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1276 at domain.example.com hanging up, cause: NORMAL_CLEARING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1276 at domain.example.com Standard HANGUP, cause: NORMAL_CLEARING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at domain.example.com) State HANGUP going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1276 at domain.example.com) State Change CS_HANGUP -> CS_REPORTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at domain.example.com) Running State Change CS_REPORTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at domain.example.com) State REPORTING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.337895 [DEBUG] mod_odbc_cdr.c:309 sql INSERT INTO cebod_cdr (used_gateway_id, phonenumber, call_uuid, leg, job_uuid, callrequest_id, accountcode, created_date, callerid, billsec, callername, starting_date, hangup_cause, duration, event_name) VALUES (68.5.94.63, 142, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, inbound, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, a17a5d15-d97e-4c70-b476-bd0ff4ac2e09, 17, 2016-02-13 09:23:26, 1276, 12, Kunal Mittal, 2016-02-13 09:23:10, NORMAL_CLEARING, 16, REQUEST_PARAMS) 2016-02-13 09:23:26.337895 [ALERT] switch_odbc.c:353 Connecting opensips 2016-02-13 09:23:26.337895 [ALERT] switch_odbc.c:397 Connected to [opensips] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1276 at domain.example.com Standard REPORTING, cause: NORMAL_CLEARING a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at domain.example.com) State REPORTING going to sleep a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1276 at domain.example.com) State Change CS_REPORTING -> CS_DESTROY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_session.c:1646 Session 1856 (sofia/internal/1276 at domain.example.com) Locked, Waiting on external entities a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [NOTICE] switch_core_session.c:1664 Session 1856 (sofia/internal/1276 at domain.example.com) Ended a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1276 at domain.example.com [CS_DESTROY] a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1276 at domain.example.com) Running State Change CS_DESTROY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at domain.example.com) State DESTROY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] mod_sofia.c:341 sofia/internal/1276 at domain.example.com SOFIA DESTROY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1276 at domain.example.com Standard DESTROY a17a5d15-d97e-4c70-b476-bd0ff4ac2e09 2016-02-13 09:23:26.347896 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at domain.example.com) State DESTROY going to sleep 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [INFO] switch_cpp.cpp:1284 Domain name is : 219.206.20.22dialed number 19499300360 destiantiotype is ivr destination is 36 account activation status: 1 2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:1356 Get Special Rate inforamtion: select billingblock, per_min_did as rate from cebod_specialrate where id_cc_card in (0,17) order by id_cc_card desc limit 1 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [INFO] switch_cpp.cpp:1284 Sending Call to IVR, IVR ID is 36 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 EXECUTE sofia/internal/+12127773456 at 66.62.60.228:5060 lua(ivr_menu.lua 36 domain.example.com) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_session.c:2781 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [APPLICATION_EXEC] 2016-02-13 09:23:26.827933 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f4164033540 Connected. 2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:1356 [ivr_menu] SQL: SELECT * FROM cebod_ivr_menus WHERE ivr_id = '36' AND ivr_menu_enabled = 'true' 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_cpp.cpp:683 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [ANSWER] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_core_media.c:373 No partner channel found, so not propagating zrtp-hash 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/+12127773456 at 66.62.60.228:5060 G729/8000 20 ms 160 samples 8000 bits 1 channels 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_codec.c:111 sofia/internal/+12127773456 at 66.62.60.228:5060 Original read codec set to G729:18 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:4485 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf send payload to 101 recv payload to 101 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/+12127773456 at 66.62.60.228:5060] 219.206.20.22 port 28274 -> 66.62.60.229 port 11570 codec: 18 ms: 20 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6332 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf send payload to 101 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6339 sofia/internal/+12127773456 at 66.62.60.228:5060 Set 2833 dtmf receive payload to 101 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_core_media.c:6362 sofia/internal/+12127773456 at 66.62.60.228:5060 Set rtp dtmf delay to 60 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/+12127773456 at 66.62.60.228:5060! 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_channel.c:3468 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change RINGING -> EARLY 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] sofia_media.c:92 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [PROGRESS_EVENT] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/+12127773456 at 66.62.60.228:5060: 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 v=0 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 o=FreeSWITCH 1455355932 1455355933 IN IP4 219.206.20.22 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 s=FreeSWITCH 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 c=IN IP4 219.206.20.22 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 t=0 0 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 m=audio 28274 RTP/AVP 18 101 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:18 G729/8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:18 annexb=no 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=rtpmap:101 telephone-event/8000 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=fmtp:101 0-16 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=ptime:20 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 a=sendrecv 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [NOTICE] switch_cpp.cpp:683 Channel [sofia/internal/+12127773456 at 66.62.60.228:5060] has been answered 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] switch_channel.c:3767 (sofia/internal/+12127773456 at 66.62.60.228:5060) Callstate Change EARLY -> ACTIVE 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [ALERT] switch_cpp.cpp:683 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [ANSWER_EVENT] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.827933 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [completed][200] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.887900 [DEBUG] sofia.c:6760 Channel sofia/internal/+12127773456 at 66.62.60.228:5060 entering state [ready][200] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:26.957827 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed. 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.837896 [DEBUG] switch_ivr_play_say.c:1359 Codec Activated L16 at 8000hz 1 channels 20ms 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.837896 [ALERT] switch_ivr_play_say.c:1290 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [AUDIO_SYNC] 022e5d6e-83a2-4397-84ba-f0f49aa6eef6 2016-02-13 09:23:27.857913 [ALERT] switch_core_io.c:1135 sofia/internal/+12127773456 at 66.62.60.228:5060 receive message [TRANSCODING_NECESSARY] 2016-02-13 09:23:27.857913 [INFO] mod_com_g729.c:126 ENCODER LICENSE ALLOCATED--->0x7f418400f440 0x7f418400f440 2016-02-13 09:23:27.857913 [INFO] mod_com_g729.c:133 ENCODER CREATED------------->0x7f418400f440 0x7f418400f440 2016-02-13 09:23:29.137959 [ALERT] switch_odbc.c:353 Connecting opensips 2016-02-13 09:23:29.137959 [ALERT] switch_odbc.c:397 Connected to [opensips] 2016-02-13 09:23:29.137959 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f424c24f000 Connected. 2016-02-13 09:23:29.157902 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f424c24f000 released. 2016-02-13 09:23:32.567895 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f424c24f000 Connected. 2016-02-13 09:23:32.567895 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f424c24f000 released. 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/124.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/164.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/638.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/474.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/290.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/128.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/608.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/428.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/374.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/17.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/236.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/322.csv 2016-02-13 09:23:34.197900 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/432.csv -------------- next part -------------- 2016-02-13 09:21:50.767867 [NOTICE] mod_logfile.c:213 New log started. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/142 at domain.example.com:5660 [03d4582f-ed06-4268-a70b-0dd9307fb4cf] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_NEW 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] sofia.c:9248 sofia/internal/142 at domain.example.com:5660 receiving invite from 73.170.102.43:49383 version: 1.6.6 64bit 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2016-02-13 09:21:55.607889 [DEBUG] sofia.c:9415 IP 73.170.102.43 Rejected by acl "domains". Falling back to Digest auth. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.607889 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/142 at domain.example.com:5660) State NEW 2016-02-13 09:21:55.607889 [DEBUG] sofia.c:2147 detaching session 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.637867 [DEBUG] sofia.c:2255 Re-attaching to session 03d4582f-ed06-4268-a70b-0dd9307fb4cf 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [DEBUG] sofia.c:9248 sofia/internal/142 at domain.example.com:5660 receiving invite from 73.170.102.43:49383 version: 1.6.6 64bit 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.647889 [ALERT] switch_core_media.c:370 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2016-02-13 09:21:55.647889 [DEBUG] sofia.c:9415 IP 73.170.102.43 Rejected by acl "domains". Falling back to Digest auth. 2016-02-13 09:21:55.647889 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [received][100] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:6770 Remote SDP: 03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0 03d4582f-ed06-4268-a70b-0dd9307fb4cf o=142 8000 8000 IN IP4 192.168.33.116 03d4582f-ed06-4268-a70b-0dd9307fb4cf s=SIP Call 03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 192.168.33.116 03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0 03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 50028 RTP/AVP 0 8 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:8 PCMA/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-15 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20 03d4582f-ed06-4268-a70b-0dd9307fb4cf 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] sofia.c:7125 (sofia/internal/142 at domain.example.com:5660) State Change CS_NEW -> CS_INIT 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_INIT 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at domain.example.com:5660) State INIT 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:88 sofia/internal/142 at domain.example.com:5660 SOFIA INIT 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:40 sofia/internal/142 at domain.example.com:5660 Standard INIT 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/142 at domain.example.com:5660) State Change CS_INIT -> CS_ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/142 at domain.example.com:5660) State INIT going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:2247 (sofia/internal/142 at domain.example.com:5660) Callstate Change DOWN -> RINGING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:141 sofia/internal/142 at domain.example.com:5660 SOFIA ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:166 sofia/internal/142 at domain.example.com:5660 Standard ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] mod_dialplan_xml.c:637 Processing 142 <142>->1276 in context public 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->limit_exceeded] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [limit_exceeded] destination_number(1276) =~ /^limit_exceeded$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_testopensip] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [from_testopensip] network_addr(73.170.102.43) =~ /^209\.216\.15\.18$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_productionopensip] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [from_productionopensip] network_addr(73.170.102.43) =~ /^209\.216\.2\.222$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [public->from_registeredusers] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [from_registeredusers] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(dialed_domain=${sip_to_host}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(domain_name=${sip_to_host}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(company_name=${sip_to_host}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(caller-id-in-from=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(sip_cid_type=none) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(Public IP is ${bind_server_ip} Domain is ${domain_name} Dialed domain is ${dialed_domain}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action lua(didhandle.lua ${destination_number}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(INFO public Local URI var_name chan var is ${domain_name} Dialed domain is ${dialed_domain}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/142 at domain.example.com:5660) State Change CS_ROUTING -> CS_EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:196 sofia/internal/142 at domain.example.com:5660 SOFIA EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:258 sofia/internal/142 at domain.example.com:5660 Standard EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(dialed_domain=domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [dialed_domain]=[domain.example.com] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(domain_name=domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [domain_name]=[domain.example.com] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(company_name=domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [company_name]=[domain.example.com] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(caller-id-in-from=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [caller-id-in-from]=[true] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(sip_cid_type=none) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_cid_type]=[none] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 log(Public IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1692 IP is 219.206.20.22 Domain is domain.example.com Dialed domain is domain.example.com 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 lua(didhandle.lua 1276) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(domain_name=domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(skip_cdr_causes=LOSE_RACE) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [skip_cdr_causes]=[LOSE_RACE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] switch_cpp.cpp:1284 RPID is dialing number is 142 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(accountcode=17) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [accountcode]=[17] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] switch_cpp.cpp:1284 String length is less that 7 should be extension 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(process_cdr=a_only) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [process_cdr]=[a_only] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_ivr.c:2085 (sofia/internal/142 at domain.example.com:5660) State Change CS_EXECUTE -> CS_ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_ivr.c:2090 sofia/internal/142 at domain.example.com:5660 receive message [TRANSFER] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [NOTICE] switch_ivr.c:2092 Transfer sofia/internal/142 at domain.example.com:5660 to XML[1276 at default] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_cpp.cpp:898 transfer result: 0 2016-02-13 09:21:55.657791 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_cpp.cpp:1103 sofia/internal/142 at domain.example.com:5660 destroy/unlink session from object 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_sofia.c:141 sofia/internal/142 at domain.example.com:5660 SOFIA ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] switch_core_state_machine.c:166 sofia/internal/142 at domain.example.com:5660 Standard ROUTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [INFO] mod_dialplan_xml.c:637 Processing 142 <142>->1276 in context default 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->longdistance] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [longdistance] destination_number(1276) =~ /^\d{10,15}$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unloop] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->freeswitch_public_conf_via_sip] continue=false 2016-02-13 09:21:55.657791 [ERR] switch_regex.c:104 COMPILE ERROR: 1 [nothing to repeat][^*9(888|8888|1616|3232)$] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(1276) =~ /^*9(888|8888|1616|3232)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->tod_example] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Date/TimeMatch (FAIL) [tod_example] break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->holiday_example] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Date/TimeMatch (FAIL) [holiday_example] break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->global-intercept] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global-intercept] destination_number(1276) =~ /^\*886$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->group-intercept] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [group-intercept] destination_number(1276) =~ /^\*8$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->intercept-ext] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [intercept-ext] destination_number(1276) =~ /^\*\*(\d+)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->redial] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [redial] destination_number(1276) =~ /^\*(redial|870)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->global] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [global] ${switch_r_sdp}(v=0 03d4582f-ed06-4268-a70b-0dd9307fb4cf o=142 8000 8000 IN IP4 192.168.33.116 03d4582f-ed06-4268-a70b-0dd9307fb4cf s=SIP Call 03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 192.168.33.116 03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0 03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 50028 RTP/AVP 0 8 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:8 PCMA/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-15 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20 03d4582f-ed06-4268-a70b-0dd9307fb4cf ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [global] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->snom-demo-2] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [snom-demo-2] destination_number(1276) =~ /^9001$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->snom-demo-1] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [snom-demo-1] destination_number(1276) =~ /^9000$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->eavesdrop] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [eavesdrop] destination_number(1276) =~ /^\*88(\d{3,5})$|^\*0(.*)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->eavesdrop] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [eavesdrop] destination_number(1276) =~ /^\*779$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call_return] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call_return] destination_number(1276) =~ /^\*69$|^\*869$|^lcr$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->del-number] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [del-number] destination_number(1276) =~ /^\*60$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->add-number] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [add-number] destination_number(1276) =~ /^\*61$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->check-number] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [check-number] destination_number(1276) =~ /^\*62$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->enable-time-condition] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [enable-time-condition] destination_number(1276) =~ /^\*63$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->disable-time-condition] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [disable-time-condition] destination_number(1276) =~ /^\*64$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->del-group] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [del-group] destination_number(1276) =~ /^\*80$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->add-group] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [add-group] destination_number(1276) =~ /^\*81$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call-group-simo] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call-group-simo] destination_number(1276) =~ /^\*82(\d{2})$|^82(\d{2})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->call-group-order] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [call-group-order] destination_number(1276) =~ /^\*83(\d{2})$|^83(\d{2})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->extension-intercom] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [extension-intercom] destination_number(1276) =~ /^\*84(\d{3,5})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->user_recodring_enabled] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [user_recodring_enabled] ${recording}(off) =~ /^(on|off)$/ break=on-true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->user_callscreening] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [user_callscreening] caller_id_number(142) =~ /^\d{10,15}$/ break=on-true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [user_callscreening] ${callscreening}(off) =~ /^(on|off)$/ break=on-true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Check IVR-based CF] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [Check IVR-based CF] destination_number(1276) =~ /^(\d+)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(dialed_number=1276) INLINE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(dialed_number=1276) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [dialed_number]=[1276] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.657791 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(cf_target=${db(select/${domain_name}-CF/1276)}) INLINE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(cf_target=) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [cf_target]=[UNDEF] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${recording}(off) =~ /^on$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{3,5}$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Check IVR-based CF] ${cf_target}() =~ /^\d{10,15}$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF from my station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF from my station] destination_number(1276) =~ /^\*72$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF cancel from my station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF cancel from my station] destination_number(1276) =~ /^\*73$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF from any station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF from any station] destination_number(1276) =~ /^\*76$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->CF cancel from any station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [CF cancel from any station] destination_number(1276) =~ /^\*77$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->NACF from my station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [NACF from my station] destination_number(1276) =~ /^\*74$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->NACF cancel from my station] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [NACF cancel from my station] destination_number(1276) =~ /^\*75$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Add_Member_To_Queue] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Add_Member_To_Queue] destination_number(1276) =~ /^\*51$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Delete_Member_From_Queue] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Delete_Member_From_Queue] destination_number(1276) =~ /^\*52$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Pickup_Call_From_Queue] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Pickup_Call_From_Queue] destination_number(1276) =~ /^\*53$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->dial-by-name] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [dial-by-name] destination_number(1276) =~ /^\*40$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->conference_withpin] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [conference_withpin] destination_number(1276) =~ /^\*3560$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Voicemail_check] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Voicemail_check] caller_id_number(142) =~ /^(1276)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->check_availability] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [check_availability] destination_number(1276) =~ /^hkjhk(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->is_check_availability] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [is_check_availability] available_to_call() =~ /true/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->osbridge] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [osbridge] destination_number(1276) =~ /(^\d{3,5})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(proxy_media=false) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(continue_on_fail=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(dialed_extension=1276) INLINE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(dialed_extension=1276) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[1276] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(domain_name=${domain_name}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(accountcode=${user_data(${dialed_extension}@${domain_name} var accountcode)}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(available_to_call=true) INLINE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(available_to_call=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [available_to_call]=[true] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action log(Extensions ${dialed_extension} is ${available_to_call}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(nacf_target=${db(select/${domain_name}-NACF/1276)}) INLINE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(nacf_target=) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [nacf_target]=[UNDEF] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${recording}(off) =~ /^on$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [osbridge] ${available_to_call}(true) =~ /^true$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(1 b s execute_extension::dx XML features) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 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)}.${uuid}.wav) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(3 b s execute_extension::cf XML features) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(4 b s execute_extension::attented_xfer XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(6 b s execute_extension::blind_xfer_to XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bind_meta_app(9 b s execute_extension::call_park XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action export(sip_contact_user=${sip_from_user}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(ringback=${us-ring}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(transfer_ringback=local_stream://moh) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(call_timeout=35) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action set(hangup_after_bridge=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action bridge(user/${dialed_extension}@${domain_name}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action sleep(1000) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{3,7}$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [osbridge] ${nacf_target}() =~ /^\d{10,15}$/ break=never 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Absolute Condition [osbridge] 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action answer() 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action sleep(500) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action voicemail(default ${domain_name} ${dialed_extension}) 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension_VM] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension_VM] destination_number(1276) =~ /^vm-(\d{3,5})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->attented_xfer_to_vm] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [attented_xfer_to_vm] destination_number(1276) =~ /^attented_xfer_to_vm$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->attented_xfer] continue=true 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [attented_xfer] destination_number(1276) =~ /^attented_xfer$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->blind_xfer_to] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [blind_xfer_to] destination_number(1276) =~ /^blind_xfer_to$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension] destination_number(1276) =~ /^disable-(1[0-9][0-9][0-9]|[1-2][0-9][0-9])$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Local_Extension_Skinny] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Local_Extension_Skinny] destination_number(1276) =~ /^(11[01][0-9])$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park-in] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park-in] destination_number(1276) =~ /^call_park$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park-out] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park-out] destination_number(1276) =~ /^\*(85\d\d)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->vmain] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [vmain] destination_number(1276) =~ /^vmain$|^\*98$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->sip_uri] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [sip_uri] destination_number(1276) =~ /^sip:(.*)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->local_uri] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [local_uri] destination_number(1276) =~ /^(local_.*)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->nb_conferences] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [nb_conferences] destination_number(1276) =~ /^\*(30\d{2,4})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->wb_conferences] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [wb_conferences] destination_number(1276) =~ /^\*(31\d{2})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->uwb_conferences] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [uwb_conferences] destination_number(1276) =~ /^\*(32\d{2})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->cdquality_conferences] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [cdquality_conferences] destination_number(1276) =~ /^\*(33\d{2})$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss_intercom] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss_intercom] destination_number(1276) =~ /^0911$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss_intercom] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss_intercom] destination_number(1276) =~ /^0912$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->mad_boss] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [mad_boss] destination_number(1276) =~ /^0913$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->queupark] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [queupark] destination_number(1276) =~ /^\*(5902)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->queueunpark] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [queueunpark] destination_number(1276) =~ /^\*(5903)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /^\*5900$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /^\*(5901)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->valet_park] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [valet_park] destination_number(1276) =~ /^(6000)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->valet_park] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [valet_park] destination_number(1276) =~ /^(60\d[1-9])$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /park\+(\d+)/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /^parking$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->park] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [park] destination_number(1276) =~ /callpark/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->unpark] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [unpark] destination_number(1276) =~ /pickup/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->wait] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [wait] destination_number(1276) =~ /^wait$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->fax_receive] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [fax_receive] destination_number(1276) =~ /^fax.*$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->fax_transmit] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [fax_transmit] destination_number(1276) =~ /^9179$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_180] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_180] destination_number(1276) =~ /^9180$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_183_uk_ring] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_183_uk_ring] destination_number(1276) =~ /^\*9181$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_183_music_ring] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_183_music_ring] destination_number(1276) =~ /^\*9182$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_post_answer_uk_ring] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(1276) =~ /^\*9183$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->ringback_post_answer_music] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [ringback_post_answer_music] destination_number(1276) =~ /^\*9184$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->show_info] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [show_info] destination_number(1276) =~ /^\*9192$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->video_record] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [video_record] destination_number(1276) =~ /^\*9193$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->video_playback] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [video_playback] destination_number(1276) =~ /^\*9194$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->delay_echo] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [delay_echo] destination_number(1276) =~ /^\*9195$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->echo] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [echo] destination_number(1276) =~ /^\*9196$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->milliwatt] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [milliwatt] destination_number(1276) =~ /^\*9197$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->tone_stream] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [tone_stream] destination_number(1276) =~ /^\*9198$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->zrtp_enrollement] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [zrtp_enrollement] destination_number(1276) =~ /^\*9787$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->hold_music] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [hold_music] destination_number(1276) =~ /^\*9664$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->laugh break] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [laugh break] destination_number(1276) =~ /^\*9386$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->101] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [101] destination_number(1276) =~ /^101$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->pizza_demo] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [pizza_demo] destination_number(1276) =~ /^(pizza|74992)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Time] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Time] destination_number(1276) =~ /^9170$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Date] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Date] destination_number(1276) =~ /^9171$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->Talking Clock Date and Time] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [Talking Clock Date and Time] destination_number(1276) =~ /^9172$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->local.example.com] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [local.example.com] ${toll_allow}() =~ /local/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->domestic.example.com] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [domestic.example.com] ${toll_allow}() =~ /domestic/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->international.example.com] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [international.example.com] ${toll_allow}() =~ /international/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->conference] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [conference] destination_number(1276) =~ /^112233$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->didforsale_ivr] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [didforsale_ivr] destination_number(1276) =~ /5566/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->domian_ivr] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (FAIL) [domian_ivr] destination_number(1276) =~ /ivr(1949.*)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 parsing [default->enum] continue=false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Regex (PASS) [enum] destination_number(1276) =~ /^(.*)$/ break=on-false 03d4582f-ed06-4268-a70b-0dd9307fb4cf Dialplan: sofia/internal/142 at domain.example.com:5660 Action transfer(1276 enum) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/142 at domain.example.com:5660) State Change CS_ROUTING -> CS_EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/142 at domain.example.com:5660) State ROUTING going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_sofia.c:196 sofia/internal/142 at domain.example.com:5660 SOFIA EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_core_state_machine.c:258 sofia/internal/142 at domain.example.com:5660 Standard EXECUTE 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-spymap/142/03d4582f-ed06-4268-a70b-0dd9307fb4cf) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-last_dial/142/1276) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 hash(insert/domain.example.com-last_dial/global/03d4582f-ed06-4268-a70b-0dd9307fb4cf) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(RFC2822_DATE=Sat, 13 Feb 2016 09:21:55 -0800) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(proxy_media=false) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [proxy_media]=[false] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(continue_on_fail=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [continue_on_fail]=[true] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(domain_name=domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [domain_name]=[domain.example.com] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 2016-02-13 09:21:55.667790 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:21:55.667790 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(accountcode=17) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [accountcode]=[17] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 log(Extensions 1276 is true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [DEBUG] mod_dptools.c:1692 1276 is true 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(1 b s execute_extension::dx XML features) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.667790 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-21-55.03d4582f-ed06-4268-a70b-0dd9307fb4cf.wav) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/domain.example.com/2016-02-13-09-21-55.03d4582f-ed06-4268-a70b-0dd9307fb4cf.wav 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(3 b s execute_extension::cf XML features) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *3 execute_extension::cf XML features 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(4 b s execute_extension::attented_xfer XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *4 execute_extension::attented_xfer XML default 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(5 b s execute_extension::attented_xfer_to_vm XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *5 execute_extension::attented_xfer_to_vm XML default 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(6 b s execute_extension::blind_xfer_to XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *6 execute_extension::blind_xfer_to XML default 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bind_meta_app(9 b s execute_extension::call_park XML default) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *9 execute_extension::call_park XML default 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 export(sip_contact_user=142) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [sip_contact_user]=[142] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(ringback=%(2000,4000,440,480)) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [ringback]=[%(2000,4000,440,480)] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(transfer_ringback=local_stream://moh) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [transfer_ringback]=[local_stream://moh] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(call_timeout=35) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [call_timeout]=[35] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 set(hangup_after_bridge=true) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] mod_dptools.c:1498 SET sofia/internal/142 at domain.example.com:5660 [hangup_after_bridge]=[true] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf EXECUTE sofia/internal/142 at domain.example.com:5660 bridge(user/1276 at domain.example.com) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [ALERT] switch_core_session.c:2781 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [caller-id-in-from]=[true] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_cid_type]=[none] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [accountcode]=[17] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [dialed_extension]=[1276] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [available_to_call]=[true] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_contact_user]=[142] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables 2016-02-13 09:21:55.677785 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:21:55.677785 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [caller-id-in-from]=[true] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_cid_type]=[none] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [skip_cdr_causes]=[LOSE_RACE] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [accountcode]=[17] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [dialed_extension]=[1276] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [available_to_call]=[true] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 13 Feb 2016 09:21:55 -0800] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [domain_name]=[domain.example.com] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_channel.c:1247 sofia/internal/142 at domain.example.com:5660 EXPORTING[export_vars] [sip_contact_user]=[142] to event 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1276 at 192.168.1.168:12113 [997ba225-1002-4fec-9b89-f7e1986ebfca] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:4776 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_NEW -> CS_INIT 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_INIT 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at 192.168.1.168:12113) State INIT 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:88 sofia/internal/1276 at 192.168.1.168:12113 SOFIA INIT 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia_glue.c:1228 sip:1276 at 68.5.94.63:12113 Setting proxy route to sofia/internal/1276 at 192.168.1.168:12113 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia_glue.c:1257 sofia/internal/1276 at 192.168.1.168:12113 sending invite version: 1.6.6 64bit 997ba225-1002-4fec-9b89-f7e1986ebfca Local SDP: 997ba225-1002-4fec-9b89-f7e1986ebfca v=0 997ba225-1002-4fec-9b89-f7e1986ebfca o=FreeSWITCH 1455351947 1455351948 IN IP4 219.206.20.22 997ba225-1002-4fec-9b89-f7e1986ebfca s=FreeSWITCH 997ba225-1002-4fec-9b89-f7e1986ebfca c=IN IP4 219.206.20.22 997ba225-1002-4fec-9b89-f7e1986ebfca t=0 0 997ba225-1002-4fec-9b89-f7e1986ebfca m=audio 32168 RTP/AVP 0 8 101 13 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:0 PCMU/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:8 PCMA/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:101 telephone-event/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=fmtp:101 0-16 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:13 CN/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=ptime:20 997ba225-1002-4fec-9b89-f7e1986ebfca a=sendrecv 997ba225-1002-4fec-9b89-f7e1986ebfca 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1276 at 192.168.1.168:12113 Standard INIT 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_INIT -> CS_ROUTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1276 at 192.168.1.168:12113) State INIT going to sleep 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_ROUTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [calling][0] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at 192.168.1.168:12113) State ROUTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] mod_sofia.c:141 sofia/internal/1276 at 192.168.1.168:12113 SOFIA ROUTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_ROUTING -> CS_CONSUME_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1276 at 192.168.1.168:12113) State ROUTING going to sleep 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_CONSUME_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1276 at 192.168.1.168:12113) State CONSUME_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1276 at 192.168.1.168:12113) State CONSUME_MEDIA going to sleep 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.677785 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 2016-02-13 09:21:55.907833 [ALERT] sofia.c:1214 sofia/internal/1276 at 192.168.1.168:12113 Same Callee ID "Outbound Call" <1276> 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [proceeding][180] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/1276 at 192.168.1.168:12113! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [DEBUG] switch_channel.c:3340 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change DOWN -> RINGING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] sofia.c:6862 sofia/internal/1276 at 192.168.1.168:12113 receive message [RING_EVENT] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:21:55.907833 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_ivr_originate.c:1216 sofia/internal/142 at domain.example.com:5660 receive message [PROGRESS] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [INFO] switch_ivr_originate.c:1216 Sending early media 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/142 at domain.example.com:5660 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_codec.c:111 sofia/internal/142 at domain.example.com:5660 Original read codec set to PCMU:0 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:4485 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf send payload to 101 recv payload to 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/142 at domain.example.com:5660] 219.206.20.22 port 30674 -> 192.168.33.116 port 50028 codec: 0 ms: 20 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6332 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf send payload to 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6339 sofia/internal/142 at domain.example.com:5660 Set 2833 dtmf receive payload to 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_media.c:6362 sofia/internal/142 at domain.example.com:5660 Set rtp dtmf delay to 60 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/142 at domain.example.com:5660! 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_channel.c:3468 (sofia/internal/142 at domain.example.com:5660) Callstate Change RINGING -> EARLY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [ALERT] sofia_media.c:92 sofia/internal/142 at domain.example.com:5660 receive message [PROGRESS_EVENT] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] mod_sofia.c:2330 Ring SDP: 03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0 03d4582f-ed06-4268-a70b-0dd9307fb4cf o=FreeSWITCH 1455353441 1455353442 IN IP4 219.206.20.22 03d4582f-ed06-4268-a70b-0dd9307fb4cf s=FreeSWITCH 03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 219.206.20.22 03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0 03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 30674 RTP/AVP 0 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-16 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=sendrecv 03d4582f-ed06-4268-a70b-0dd9307fb4cf 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_core_codec.c:221 sofia/internal/142 at domain.example.com:5660 Push codec L16:100 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.927830 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [early][183] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [ALERT] switch_channel.c:3507 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:55.947831 [ALERT] switch_core_io.c:1135 sofia/internal/142 at domain.example.com:5660 receive message [TRANSCODING_NECESSARY] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:21:56.817831 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.33.116:50028 to 73.170.102.43:50028 2016-02-13 09:22:00.847852 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:22:00.847852 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 2016-02-13 09:22:03.957850 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:22:03.957850 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 2016-02-13 09:22:04.037826 [ALERT] sofia.c:1214 sofia/internal/1276 at 192.168.1.168:12113 Same Callee ID "Outbound Call" <1276> 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [completing][200] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6770 Remote SDP: 997ba225-1002-4fec-9b89-f7e1986ebfca v=0 997ba225-1002-4fec-9b89-f7e1986ebfca o=1276 82204 1 IN IP4 192.168.1.168 997ba225-1002-4fec-9b89-f7e1986ebfca s=- 997ba225-1002-4fec-9b89-f7e1986ebfca c=IN IP4 192.168.1.168 997ba225-1002-4fec-9b89-f7e1986ebfca t=0 0 997ba225-1002-4fec-9b89-f7e1986ebfca m=audio 8000 RTP/AVP 0 96 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:0 PCMU/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=rtpmap:96 telephone-event/8000 997ba225-1002-4fec-9b89-f7e1986ebfca a=ptime:20 997ba225-1002-4fec-9b89-f7e1986ebfca 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] sofia.c:6760 Channel sofia/internal/1276 at 192.168.1.168:12113 entering state [ready][200] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:413 Looking for zrtp-hash 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:368 Deciding whether to pass zrtp-hash between legs 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:376 Found peer channel; propagating zrtp-hash if set 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_media.c:303 Deciding whether to pass zrtp-hash between a-leg and b-leg 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 96 at 8000 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1276 at 192.168.1.168:12113 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_codec.c:111 sofia/internal/1276 at 192.168.1.168:12113 Original read codec set to PCMU:0 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 96 at 8000 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:4485 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf send payload to 96 recv payload to 101 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1276 at 192.168.1.168:12113] 219.206.20.22 port 32168 -> 192.168.1.168 port 8000 codec: 0 ms: 20 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6332 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf send payload to 96 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6339 sofia/internal/1276 at 192.168.1.168:12113 Set 2833 dtmf receive payload to 101 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_core_media.c:6362 sofia/internal/1276 at 192.168.1.168:12113 Set rtp dtmf delay to 60 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [NOTICE] sofia.c:7724 Channel [sofia/internal/1276 at 192.168.1.168:12113] has been answered 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [DEBUG] switch_channel.c:3767 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change RINGING -> ACTIVE 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] sofia.c:7724 sofia/internal/1276 at 192.168.1.168:12113 receive message [ANSWER_EVENT] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.037826 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_core_codec.c:246 sofia/internal/142 at domain.example.com:5660 Restore previous codec PCMU:0. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3550 sofia/internal/142 at domain.example.com:5660 receive message [ANSWER] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/142 at domain.example.com:5660: 03d4582f-ed06-4268-a70b-0dd9307fb4cf v=0 03d4582f-ed06-4268-a70b-0dd9307fb4cf o=FreeSWITCH 1455353441 1455353443 IN IP4 219.206.20.22 03d4582f-ed06-4268-a70b-0dd9307fb4cf s=FreeSWITCH 03d4582f-ed06-4268-a70b-0dd9307fb4cf c=IN IP4 219.206.20.22 03d4582f-ed06-4268-a70b-0dd9307fb4cf t=0 0 03d4582f-ed06-4268-a70b-0dd9307fb4cf m=audio 30674 RTP/AVP 0 101 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:0 PCMU/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=rtpmap:101 telephone-event/8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=fmtp:101 0-16 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=ptime:20 03d4582f-ed06-4268-a70b-0dd9307fb4cf a=sendrecv 03d4582f-ed06-4268-a70b-0dd9307fb4cf 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/142 at domain.example.com:5660] has been answered 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_channel.c:3767 (sofia/internal/142 at domain.example.com:5660) Callstate Change EARLY -> ACTIVE 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3550 sofia/internal/142 at domain.example.com:5660 receive message [ANSWER_EVENT] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [completed][200] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1276 at 192.168.1.168:12113] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3944 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3941 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1276 at 192.168.1.168:12113] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3944 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_originate.c:3941 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1452 sofia/internal/1276 at 192.168.1.168:12113 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:590 sofia/internal/1276 at 192.168.1.168:12113 session thread sleep state: CS_CONSUME_MEDIA! 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr.c:217 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1451 sofia/internal/142 at domain.example.com:5660 receive message [AUDIO_SYNC] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1551 sofia/internal/1276 at 192.168.1.168:12113 receive message [BRIDGE] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_media.c:9156 sofia/internal/1276 at 192.168.1.168:12113 PAUSE Jitterbuffer 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [ALERT] switch_ivr_bridge.c:1559 sofia/internal/142 at domain.example.com:5660 receive message [BRIDGE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.047852 [DEBUG] switch_core_media.c:9156 sofia/internal/142 at domain.example.com:5660 PAUSE Jitterbuffer 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [ALERT] switch_core_state_machine.c:594 sofia/internal/1276 at 192.168.1.168:12113 session thread wake state: CS_CONSUME_MEDIA! 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_EXCHANGE_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1276 at 192.168.1.168:12113) State EXCHANGE_MEDIA 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.047852 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.067829 [ALERT] switch_core_io.c:420 sofia/internal/142 at domain.example.com:5660 receive message [TRANSCODING_NECESSARY] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.087830 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed. 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.257832 [INFO] switch_rtp.c:6616 Auto Changing audio port from 192.168.1.168:8000 to 68.5.94.63:8000 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.587826 [DEBUG] sofia.c:6760 Channel sofia/internal/142 at domain.example.com:5660 entering state [ready][200] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:04.597832 [ALERT] switch_ivr_bridge.c:283 sofia/internal/1276 at 192.168.1.168:12113 receive message [DISPLAY] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:04.607833 [ALERT] switch_ivr_bridge.c:283 sofia/internal/142 at domain.example.com:5660 receive message [DISPLAY] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:06.197829 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53854 but expected: 53852 lost: 2 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:06.617854 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53866 but expected: 53865 lost: 1 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:08.737851 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53961 but expected: 53960 lost: 1 2016-02-13 09:22:08.827825 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:22:08.827825 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:10.097827 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53977 but expected: 53974 lost: 3 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:10.637831 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 53990 but expected: 53989 lost: 1 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:11.717833 [ALERT] switch_rtp.c:1612 sofia/internal/1276 at 192.168.1.168:12113 Got: audio seq 54036 but expected: 54035 lost: 1 2016-02-13 09:22:11.967873 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f41ac03e390 Connected. 2016-02-13 09:22:11.967873 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f41ac03e390 released. 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.107862 [NOTICE] sofia.c:952 Hangup sofia/internal/142 at domain.example.com:5660 [CS_EXECUTE] [NORMAL_CLEARING] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.107862 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/142 at domain.example.com:5660] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/142 at domain.example.com:5660 ending bridge by request from write function 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1276 at 192.168.1.168:12113] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/1276 at 192.168.1.168:12113 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1276 at 192.168.1.168:12113) State EXCHANGE_MEDIA going to sleep 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_HANGUP 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1276 at 192.168.1.168:12113) Callstate Change ACTIVE -> HANGUP 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at 192.168.1.168:12113) State HANGUP 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:425 sofia/internal/1276 at 192.168.1.168:12113 Overriding SIP cause 480 with 200 from the other leg 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1276 at 192.168.1.168:12113 hanging up, cause: NORMAL_CLEARING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/1276 at 192.168.1.168:12113 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1276 at 192.168.1.168:12113 Standard HANGUP, cause: NORMAL_CLEARING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1276 at 192.168.1.168:12113) State HANGUP going to sleep 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [ALERT] switch_ivr_bridge.c:1689 sofia/internal/1276 at 192.168.1.168:12113 receive message [UNBRIDGE] 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:1689 sofia/internal/1276 at 192.168.1.168:12113 skip receive message [UNBRIDGE] (channel is hungup already) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [ALERT] switch_ivr_bridge.c:1692 sofia/internal/142 at domain.example.com:5660 receive message [UNBRIDGE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/142 at domain.example.com:5660 skip receive message [UNBRIDGE] (channel is hungup already) 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_HANGUP -> CS_REPORTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_REPORTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [ALERT] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 receive message [APPLICATION_EXEC_COMPLETE] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_session.c:2796 sofia/internal/142 at domain.example.com:5660 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/142 at domain.example.com:5660) State EXECUTE going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_HANGUP 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at 192.168.1.168:12113) State REPORTING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1276 at 192.168.1.168:12113 Standard REPORTING, cause: NORMAL_CLEARING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1276 at 192.168.1.168:12113) State REPORTING going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/142 at domain.example.com:5660) Callstate Change ACTIVE -> HANGUP 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1276 at 192.168.1.168:12113) State Change CS_REPORTING -> CS_DESTROY 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_session.c:1646 Session 1855 (sofia/internal/1276 at 192.168.1.168:12113) Locked, Waiting on external entities 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_core_session.c:1664 Session 1855 (sofia/internal/1276 at 192.168.1.168:12113) Ended 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1276 at 192.168.1.168:12113 [CS_DESTROY] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at domain.example.com:5660) State HANGUP 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:431 Channel sofia/internal/142 at domain.example.com:5660 hanging up, cause: NORMAL_CLEARING 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1276 at 192.168.1.168:12113) Running State Change CS_DESTROY 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at 192.168.1.168:12113) State DESTROY 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] mod_sofia.c:341 sofia/internal/1276 at 192.168.1.168:12113 SOFIA DESTROY 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1276 at 192.168.1.168:12113 Standard DESTROY 997ba225-1002-4fec-9b89-f7e1986ebfca 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1276 at 192.168.1.168:12113) State DESTROY going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:60 sofia/internal/142 at domain.example.com:5660 Standard HANGUP, cause: NORMAL_CLEARING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/142 at domain.example.com:5660) State HANGUP going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/142 at domain.example.com:5660) State Change CS_HANGUP -> CS_REPORTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_REPORTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at domain.example.com:5660) State REPORTING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.117826 [DEBUG] mod_odbc_cdr.c:309 sql INSERT INTO cebod_cdr (used_gateway_id, phonenumber, call_uuid, leg, job_uuid, callrequest_id, accountcode, created_date, callerid, billsec, callername, starting_date, hangup_cause, duration, event_name) VALUES (73.170.102.43, 1276, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, inbound, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, 03d4582f-ed06-4268-a70b-0dd9307fb4cf, 17, 2016-02-13 09:22:12, 142, 8, 142, 2016-02-13 09:21:55, NORMAL_CLEARING, 17, REQUEST_PARAMS) 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:104 sofia/internal/142 at domain.example.com:5660 Standard REPORTING, cause: NORMAL_CLEARING 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/142 at domain.example.com:5660) State REPORTING going to sleep 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/142 at domain.example.com:5660) State Change CS_REPORTING -> CS_DESTROY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_session.c:1646 Session 1854 (sofia/internal/142 at domain.example.com:5660) Locked, Waiting on external entities 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [NOTICE] switch_core_session.c:1664 Session 1854 (sofia/internal/142 at domain.example.com:5660) Ended 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/142 at domain.example.com:5660 [CS_DESTROY] 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/142 at domain.example.com:5660) Running State Change CS_DESTROY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at domain.example.com:5660) State DESTROY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] mod_sofia.c:341 sofia/internal/142 at domain.example.com:5660 SOFIA DESTROY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:111 sofia/internal/142 at domain.example.com:5660 Standard DESTROY 03d4582f-ed06-4268-a70b-0dd9307fb4cf 2016-02-13 09:22:12.127828 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/142 at domain.example.com:5660) State DESTROY going to sleep From rajil.s at gmail.com Sun Feb 14 05:53:54 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Sat, 13 Feb 2016 20:53:54 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode Message-ID: Hello, I have a remote sip phone (Linksys SPA3102) which only supports PCMU. When I call to this remote sip phone i get a 406 error that opus is not supported as shown by the sip trace below. However, if I force the codec to absolute like this {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 the call works fine. Is there anyway I can make FreeSWITCH to automatically transcode without forcing the codec string in the dial plan? The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and outbound_codec_prefs=PCMU,PCMA,GSM ---------------------------siptrace-------------------------------- recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: ------------------------------------------------------------------------ SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 From: "202" ;tag=DFX0FUvr2vNcm To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q CSeq: 87372504 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: ------------------------------------------------------------------------ ACK sip:303 at 192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej Max-Forwards: 68 From: "202" ;tag=DFX0FUvr2vNcm To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 CSeq: 87372504 ACK Content-Length: 0 ------------------------------------------------------------------------ 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel sofia/internal/303 at 192.168.1.5 entering state [terminated][406] 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Thanks Rajil From dave at dchorton.com Sun Feb 14 06:49:04 2016 From: dave at dchorton.com (Dave Horton) Date: Sat, 13 Feb 2016 22:49:04 -0500 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: References: Message-ID: <67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com> Redo with sofia global siptrace on so we can see the SIP messaging On Feb 13, 2016, at 12:45 PM, 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 > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/ff31df46/attachment-0001.html From vbvbrj at gmail.com Sun Feb 14 10:35:07 2016 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 14 Feb 2016 09:35:07 +0200 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: References: <56B75BDE.6020604@gmail.com> Message-ID: <56C02E2B.2090304@gmail.com> On 14.02.2016 02:38, Abdul Hakeem wrote: > Hello Mimiko, > Did you find any resolution to the issue ? > Cheers, > AH No, no answers to my questions. I use start-up script in lua.conf and bind to event in order to avoid segfault. -- Mimiko desu. From idokan at gmail.com Sun Feb 14 17:16:44 2016 From: idokan at gmail.com (ik) Date: Sun, 14 Feb 2016 16:16:44 +0200 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: <56C02E2B.2090304@gmail.com> References: <56B75BDE.6020604@gmail.com> <56C02E2B.2090304@gmail.com> Message-ID: I'ved created a Perl daemon for capturing specific events. You can filter to do so. Ido On Feb 14, 2016 9:37 AM, "Mimiko" wrote: > On 14.02.2016 02:38, Abdul Hakeem wrote: > > Hello Mimiko, > > Did you find any resolution to the issue ? > > Cheers, > > AH > > > No, no answers to my questions. > > I use start-up script in lua.conf and bind to event in order to avoid > segfault. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160214/019e1ce8/attachment.html From krice at freeswitch.org Sun Feb 14 18:44:21 2016 From: krice at freeswitch.org (Ken Rice) Date: Sun, 14 Feb 2016 09:44:21 -0600 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: <56C02E2B.2090304@gmail.com> References: <56B75BDE.6020604@gmail.com> <56C02E2B.2090304@gmail.com> Message-ID: <7ee401d1673e$93284c20$b978e460$@freeswitch.org> Is there a ticket open on this segfault you are avoiding and how to reproduce it? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko Sent: Sunday, February 14, 2016 1:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? On 14.02.2016 02:38, Abdul Hakeem wrote: > Hello Mimiko, > Did you find any resolution to the issue ? > Cheers, > AH No, no answers to my questions. I use start-up script in lua.conf and bind to event in order to avoid segfault. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 lists at kavun.ch Sun Feb 14 19:18:16 2016 From: lists at kavun.ch (Emrah) Date: Sun, 14 Feb 2016 17:18:16 +0100 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> Message-ID: <1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch> I love all your ideas, and thanks for sharing. The best option remains to test things out in real scenarios and see what happens. I?m already distributing my media traffic. Now I?d like to make the SIP part redundant. Just got some server resources in different locations. I?ll report back with my findings! E > On Feb 12, 2016, at 10:09 AM, Andrew Cassidy 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/20160214/5ccfb228/attachment-0001.html From abaci64 at gmail.com Sun Feb 14 19:41:47 2016 From: abaci64 at gmail.com (Abaci B) Date: Sun, 14 Feb 2016 11:41:47 -0500 Subject: [Freeswitch-users] AMD Message-ID: Hi all, I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls (doing lua IVR with no transcoding) and conferencing. Any help, feedback, benchmarks or personal experience would be appreciated Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160214/464b90ac/attachment.html From gmaruzz at gmail.com Sun Feb 14 20:28:54 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 14 Feb 2016 18:28:54 +0100 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: How you originate the call? Is a bridge? From which phone? Also, please pastebin the complete sip trace (from start of leg A to end of both legs) and put here a link to pastebin Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: > Hello, > > I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > When I call to this remote sip phone i get a 406 error that opus is > not supported as shown by the sip trace below. However, if I force the > codec to absolute like this > {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > the call works fine. > > Is there anyway I can make FreeSWITCH to automatically transcode > without forcing the codec string in the dial plan? > > The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and > outbound_codec_prefs=PCMU,PCMA,GSM > > ---------------------------siptrace-------------------------------- > > recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > ------------------------------------------------------------------------ > SIP/2.0 406 Not Acceptable > Via: SIP/2.0/UDP > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > From: "202" ;tag=DFX0FUvr2vNcm > To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > CSeq: 87372504 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > ------------------------------------------------------------------------ > ACK sip:303 at 192.168.1.5 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > Max-Forwards: 68 > From: "202" ;tag=DFX0FUvr2vNcm > To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > CSeq: 87372504 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > [SERVICE_NOT_IMPLEMENTED] > > Thanks > Rajil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160214/4e197b79/attachment.html From vbvbrj at gmail.com Sun Feb 14 20:37:04 2016 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 14 Feb 2016 19:37:04 +0200 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: <7ee401d1673e$93284c20$b978e460$@freeswitch.org> References: <56B75BDE.6020604@gmail.com> <56C02E2B.2090304@gmail.com> <7ee401d1673e$93284c20$b978e460$@freeswitch.org> Message-ID: <56C0BB40.3060909@gmail.com> On 14.02.2016 17:44, Ken Rice wrote: > Is there a ticket open on this segfault you are avoiding and how to > reproduce it? No, I didn't open any ticket, because my FS version is 1.3.13b cad607d72e compiled at 2013.01.18, so the bug may be fixed or may be not. I can't update right now the version. My main question is what is best to capture event? 1) Using in lua.conf. 2) or using in lua.conf and hook to event like for e in (function() return con:pop(1) end) do.... Comparing performance and stability. Using 1st method has the advantage that I can change script without restarting FS. But at my version running it for several hours FS stops with segfault. If using 1st method to capture events is better, then I'll update FS and see if segfault persists or not. If 2nd method is better - then there is no need for me to update FS right now. To reproduce see my first message on this. -- Mimiko desu. From vbvbrj at gmail.com Sun Feb 14 20:39:46 2016 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 14 Feb 2016 19:39:46 +0200 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: References: <56B75BDE.6020604@gmail.com> <56C02E2B.2090304@gmail.com> Message-ID: <56C0BBE2.5050206@gmail.com> On 14.02.2016 16:16, ik wrote: > I'ved created a Perl daemon for capturing specific events. > > You can filter to do so. > Do you capture all events in your perl script? Do you need to restart perl script to capture new events after modifying the script? If think you do and while perl script is restarting events may be missed. I prefer to use lua, because it is integrated in FS code and is much faster. -- Mimiko desu. From mike at jerris.com Sun Feb 14 20:52:21 2016 From: mike at jerris.com (Michael Jerris) Date: Sun, 14 Feb 2016 12:52:21 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: You would have to test to see, most of my experience is a couple chip generations back now but bang for the buck was much better with intel then and we saw many more weird issues on AMD boxes. On Sunday, February 14, 2016, Abaci B wrote: > Hi all, > I know that in general Intel Xeon processors are recommended over AMD for > FreeSWITCH, my question is if AMD is really bad or just not as good as > Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about > the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I > go for the AMD which will give me 4x the amount of cores or stick to Xeon, > my goal is to get the maximum amount of channels/calls (doing lua IVR with > no transcoding) and conferencing. > Any help, feedback, benchmarks or personal experience would be appreciated > Thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160214/6f9d6a9e/attachment.html From rajil.s at gmail.com Sun Feb 14 22:04:32 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Sun, 14 Feb 2016 13:04:32 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: The siptrace is at http://pastebin.com/xiGqtj1Y The call is being made from 303 (Android/CSipsimple with OPUS codec) to 208 (pjsua test client with PCMU codec). The error is on line 545. On 14 February 2016 at 11:28, Giovanni Maruzzelli wrote: > How you originate the call? Is a bridge? From which phone? > > Also, please pastebin the complete sip trace (from start of leg A to end of > both legs) and put here a link to pastebin > > Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: >> >> Hello, >> >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. >> When I call to this remote sip phone i get a 406 error that opus is >> not supported as shown by the sip trace below. However, if I force the >> codec to absolute like this >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >> the call works fine. >> >> Is there anyway I can make FreeSWITCH to automatically transcode >> without forcing the codec string in the dial plan? >> >> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and >> outbound_codec_prefs=PCMU,PCMA,GSM >> >> ---------------------------siptrace-------------------------------- >> >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >> >> ------------------------------------------------------------------------ >> SIP/2.0 406 Not Acceptable >> Via: SIP/2.0/UDP >> >> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> From: "202" ;tag=DFX0FUvr2vNcm >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> CSeq: 87372504 INVITE >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >> >> ------------------------------------------------------------------------ >> ACK sip:303 at 192.168.1.5 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >> Max-Forwards: 68 >> From: "202" ;tag=DFX0FUvr2vNcm >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> CSeq: 87372504 ACK >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >> [SERVICE_NOT_IMPLEMENTED] >> >> Thanks >> Rajil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 rtreleaven at bunnykick.ca Sun Feb 14 22:33:49 2016 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Sun, 14 Feb 2016 14:33:49 -0500 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: Look at the invite starting at 480. You are only offering opus to the callee. On Feb 14, 2016 2:05 PM, "Rajil Saraswat" wrote: > The siptrace is at http://pastebin.com/xiGqtj1Y > > The call is being made from 303 (Android/CSipsimple with OPUS codec) > to 208 (pjsua test client with PCMU codec). The error is on line 545. > > On 14 February 2016 at 11:28, Giovanni Maruzzelli > wrote: > > How you originate the call? Is a bridge? From which phone? > > > > Also, please pastebin the complete sip trace (from start of leg A to end > of > > both legs) and put here a link to pastebin > > > > Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: > >> > >> Hello, > >> > >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >> When I call to this remote sip phone i get a 406 error that opus is > >> not supported as shown by the sip trace below. However, if I force the > >> codec to absolute like this > >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > >> the call works fine. > >> > >> Is there anyway I can make FreeSWITCH to automatically transcode > >> without forcing the codec string in the dial plan? > >> > >> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and > >> outbound_codec_prefs=PCMU,PCMA,GSM > >> > >> ---------------------------siptrace-------------------------------- > >> > >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >> > >> ------------------------------------------------------------------------ > >> SIP/2.0 406 Not Acceptable > >> Via: SIP/2.0/UDP > >> > >> > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >> From: "202" ;tag=DFX0FUvr2vNcm > >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >> CSeq: 87372504 INVITE > >> Content-Length: 0 > >> > >> > >> ------------------------------------------------------------------------ > >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >> > >> ------------------------------------------------------------------------ > >> ACK sip:303 at 192.168.1.5 SIP/2.0 > >> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >> Max-Forwards: 68 > >> From: "202" ;tag=DFX0FUvr2vNcm > >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >> CSeq: 87372504 ACK > >> Content-Length: 0 > >> > >> > >> ------------------------------------------------------------------------ > >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >> [SERVICE_NOT_IMPLEMENTED] > >> > >> Thanks > >> Rajil > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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/20160214/ef245949/attachment.html From rtreleaven at bunnykick.ca Sun Feb 14 22:56:56 2016 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Sun, 14 Feb 2016 14:56:56 -0500 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat wrote: > The siptrace is at http://pastebin.com/xiGqtj1Y > > The call is being made from 303 (Android/CSipsimple with OPUS codec) > to 208 (pjsua test client with PCMU codec). The error is on line 545. > > On 14 February 2016 at 11:28, Giovanni Maruzzelli > wrote: > > How you originate the call? Is a bridge? From which phone? > > > > Also, please pastebin the complete sip trace (from start of leg A to end > of > > both legs) and put here a link to pastebin > > > > Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: > >> > >> Hello, > >> > >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >> When I call to this remote sip phone i get a 406 error that opus is > >> not supported as shown by the sip trace below. However, if I force the > >> codec to absolute like this > >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > >> the call works fine. > >> > >> Is there anyway I can make FreeSWITCH to automatically transcode > >> without forcing the codec string in the dial plan? > >> > >> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and > >> outbound_codec_prefs=PCMU,PCMA,GSM > >> > >> ---------------------------siptrace-------------------------------- > >> > >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >> > >> ------------------------------------------------------------------------ > >> SIP/2.0 406 Not Acceptable > >> Via: SIP/2.0/UDP > >> > >> > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >> From: "202" ;tag=DFX0FUvr2vNcm > >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >> CSeq: 87372504 INVITE > >> Content-Length: 0 > >> > >> > >> ------------------------------------------------------------------------ > >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >> > >> ------------------------------------------------------------------------ > >> ACK sip:303 at 192.168.1.5 SIP/2.0 > >> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >> Max-Forwards: 68 > >> From: "202" ;tag=DFX0FUvr2vNcm > >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >> CSeq: 87372504 ACK > >> Content-Length: 0 > >> > >> > >> ------------------------------------------------------------------------ > >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >> [SERVICE_NOT_IMPLEMENTED] > >> > >> Thanks > >> Rajil > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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/20160214/169c1459/attachment-0001.html From rajil.s at gmail.com Mon Feb 15 00:37:47 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Sun, 14 Feb 2016 15:37:47 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: Thanks, after setting media_mix_inbound_outbound_codecs=true, transcoding happens automatically. I remember not setting this variable in other installations and transcoding used to work out of the box. Is media_mix_inbound_outbound_codecs=true default in Freeswitch? On 14 February 2016 at 13:56, Russell Treleaven wrote: > fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > > On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat wrote: >> >> The siptrace is at http://pastebin.com/xiGqtj1Y >> >> The call is being made from 303 (Android/CSipsimple with OPUS codec) >> to 208 (pjsua test client with PCMU codec). The error is on line 545. >> >> On 14 February 2016 at 11:28, Giovanni Maruzzelli >> wrote: >> > How you originate the call? Is a bridge? From which phone? >> > >> > Also, please pastebin the complete sip trace (from start of leg A to end >> > of >> > both legs) and put here a link to pastebin >> > >> > Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: >> >> >> >> Hello, >> >> >> >> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. >> >> When I call to this remote sip phone i get a 406 error that opus is >> >> not supported as shown by the sip trace below. However, if I force the >> >> codec to absolute like this >> >> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >> >> the call works fine. >> >> >> >> Is there anyway I can make FreeSWITCH to automatically transcode >> >> without forcing the codec string in the dial plan? >> >> >> >> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and >> >> outbound_codec_prefs=PCMU,PCMA,GSM >> >> >> >> ---------------------------siptrace-------------------------------- >> >> >> >> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> SIP/2.0 406 Not Acceptable >> >> Via: SIP/2.0/UDP >> >> >> >> >> >> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >> >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> >> From: "202" ;tag=DFX0FUvr2vNcm >> >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> >> CSeq: 87372504 INVITE >> >> Content-Length: 0 >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> ACK sip:303 at 192.168.1.5 SIP/2.0 >> >> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >> >> Max-Forwards: 68 >> >> From: "202" ;tag=DFX0FUvr2vNcm >> >> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> >> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> >> CSeq: 87372504 ACK >> >> Content-Length: 0 >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >> >> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >> >> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >> >> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >> >> [SERVICE_NOT_IMPLEMENTED] >> >> >> >> Thanks >> >> Rajil >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 krice at freeswitch.org Mon Feb 15 01:07:14 2016 From: krice at freeswitch.org (Ken Rice) Date: Sun, 14 Feb 2016 16:07:14 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it Sent from my iPhone > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > transcoding happens automatically. I remember not setting this > variable in other installations and transcoding used to work out of > the box. Is media_mix_inbound_outbound_codecs=true default in > Freeswitch? > >> On 14 February 2016 at 13:56, Russell Treleaven wrote: >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat wrote: >>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y >>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli >>> wrote: >>>> How you originate the call? Is a bridge? From which phone? >>>> >>>> Also, please pastebin the complete sip trace (from start of leg A to end >>>> of >>>> both legs) and put here a link to pastebin >>>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: >>>>> >>>>> Hello, >>>>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. >>>>> When I call to this remote sip phone i get a 406 error that opus is >>>>> not supported as shown by the sip trace below. However, if I force the >>>>> codec to absolute like this >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >>>>> the call works fine. >>>>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode >>>>> without forcing the codec string in the dial plan? >>>>> >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and >>>>> outbound_codec_prefs=PCMU,PCMA,GSM >>>>> >>>>> ---------------------------siptrace-------------------------------- >>>>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 406 Not Acceptable >>>>> Via: SIP/2.0/UDP >>>>> >>>>> >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> CSeq: 87372504 INVITE >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >>>>> Max-Forwards: 68 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> CSeq: 87372504 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >>>>> [SERVICE_NOT_IMPLEMENTED] >>>>> >>>>> Thanks >>>>> Rajil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 colton.conor at gmail.com Mon Feb 15 03:20:39 2016 From: colton.conor at gmail.com (Colton Conor) Date: Sun, 14 Feb 2016 18:20:39 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> Message-ID: So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. Plus if you use TLS for encryption and security then you are already using TCP right? On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: > This behavior changed a while ago. This was dictates by ever growing SDPs > and exceeding MTUs causing udp fragmentation. Udp does not deal with > fragmentation and everyone refuses to fully implement sip over tcp for some > reason even tho a ton of things support it and the RFCs require it > > Sent from my iPhone > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven < > rtreleaven at bunnykick.ca> wrote: > >> fyi > https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) > >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> wrote: > >>>> How you originate the call? Is a bridge? From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A to > end > >>>> of > >>>> both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha > scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus is > >>>>> not supported as shown by the sip trace below. However, if I force > the > >>>>> codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA > and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace-------------------------------- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at 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/20160214/8fef2668/attachment-0001.html From idokan at gmail.com Mon Feb 15 10:41:12 2016 From: idokan at gmail.com (ik) Date: Mon, 15 Feb 2016 09:41:12 +0200 Subject: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events? In-Reply-To: <56C0BBE2.5050206@gmail.com> References: <56B75BDE.6020604@gmail.com> <56C02E2B.2090304@gmail.com> <56C0BBE2.5050206@gmail.com> Message-ID: On my case, I listen only for specific events, so I use the filter api call, but without it, you will get all events. On Feb 14, 2016 7:41 PM, "Mimiko" wrote: > On 14.02.2016 16:16, ik wrote: > > I'ved created a Perl daemon for capturing specific events. > > > > You can filter to do so. > > > > Do you capture all events in your perl script? > > Do you need to restart perl script to capture new events after modifying > the script? If think you do and while perl script is restarting events > may be missed. > > I prefer to use lua, because it is integrated in FS code and is much > faster. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/5a23ddd7/attachment.html From stefano.favaro at edistar.com Mon Feb 15 12:12:25 2016 From: stefano.favaro at edistar.com (Stefano Favaro) Date: Mon, 15 Feb 2016 10:12:25 +0100 (CET) Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: Message-ID: <1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com> Thank you Italo. This is my log https://pastebin.freeswitch.org/24555 and this is the id of my user spy call 94bae5a0-dd2c-4d0e-99fd-8ef81e753227 Stefano ----- Messaggio originale ----- Da: "?talo Rossi" A: "FreeSWITCH Users Help" Inviato: Gioved?, 11 febbraio 2016 18:52:46 Oggetto: Re: [Freeswitch-users] Problem with mod_spy 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 < stefano.favaro at edistar.com > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160215/c459e2f7/attachment.html From matt at supportedbusiness.com Mon Feb 15 17:28:11 2016 From: matt at supportedbusiness.com (Matt Broad) Date: Mon, 15 Feb 2016 14:28:11 +0000 Subject: [Freeswitch-users] bind_digit_action in a conference Message-ID: Hi, is it possible to bind a digit action and set the target as another member of a conference? I would like to be able to press a digit and have another member execute an extension where they enter some digits before returning back to the conference. thanks Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/921fc966/attachment.html From adnan.ahmed1 at gmail.com Mon Feb 15 03:10:02 2016 From: adnan.ahmed1 at gmail.com (Adnan Ahmed) Date: Mon, 15 Feb 2016 00:10:02 +0000 Subject: [Freeswitch-users] FS Bridged call, no RTP until DTMF pressed In-Reply-To: References: Message-ID: You can disregard this message. Further testing and isolation of the software/hardware found the cause to be in the chan_dahdi.c file for asterisk. (Not actually an error but it was what was causing me issues.) The conf mute was being turned on because it thought it was receiving a DTMFdown event on the E&m wink with Feature Group B MF signalling trunk. Not sure if it was an error or anything, so I just hard coded the file to not use confmute, and now it works fine. Adnan. On Thu, Feb 11, 2016, 09:32 Adnan Ahmed wrote: > 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/20160215/1fee8520/attachment.html From silviu.cpp at gmail.com Mon Feb 15 17:34:52 2016 From: silviu.cpp at gmail.com (Caragea Silviu) Date: Mon, 15 Feb 2016 16:34:52 +0200 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS Message-ID: Hello, Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues for several modules Silviu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/3e4ac9b3/attachment-0001.html From gmaruzz at gmail.com Mon Feb 15 18:22:37 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 15 Feb 2016 16:22:37 +0100 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: At this moment in time, Debian 8 (Jessie) is the recommended and supported platform. Il 15/Feb/2016 15:36, "Caragea Silviu" ha scritto: > Hello, > > Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues > for several modules > > Silviu > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/05615998/attachment.html From mike at jerris.com Mon Feb 15 18:33:46 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Feb 2016 10:33:46 -0500 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> Message-ID: any device that even remotely follows sip specs supports TCP. Most phones I have seen do. On Sunday, February 14, 2016, Colton Conor wrote: > So is TCP the preferred method of doing SIP these days? I like TCP with > endpoints as they always break through firewalls and we never seem to have > in issue with TCP. However UDP is a headache. So if you have the choice why > not do TCP? I realize some devices only support UDP, but the majority of > SIP phones out there today do support TCP. > > Plus if you use TLS for encryption and security then you are already using > TCP right? > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice > wrote: > >> This behavior changed a while ago. This was dictates by ever growing SDPs >> and exceeding MTUs causing udp fragmentation. Udp does not deal with >> fragmentation and everyone refuses to fully implement sip over tcp for some >> reason even tho a ton of things support it and the RFCs require it >> >> Sent from my iPhone >> >> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat > > wrote: >> > >> > Thanks, after setting media_mix_inbound_outbound_codecs=true, >> > transcoding happens automatically. I remember not setting this >> > variable in other installations and transcoding used to work out of >> > the box. Is media_mix_inbound_outbound_codecs=true default in >> > Freeswitch? >> > >> >> On 14 February 2016 at 13:56, Russell Treleaven < >> rtreleaven at bunnykick.ca >> > wrote: >> >> fyi >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >> >> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > > wrote: >> >>> >> >>> The siptrace is at http://pastebin.com/xiGqtj1Y >> >>> >> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >> >>> >> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > > >> >>> wrote: >> >>>> How you originate the call? Is a bridge? From which phone? >> >>>> >> >>>> Also, please pastebin the complete sip trace (from start of leg A to >> end >> >>>> of >> >>>> both legs) and put here a link to pastebin >> >>>> >> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" > > ha scritto: >> >>>>> >> >>>>> Hello, >> >>>>> >> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports >> PCMU. >> >>>>> When I call to this remote sip phone i get a 406 error that opus is >> >>>>> not supported as shown by the sip trace below. However, if I force >> the >> >>>>> codec to absolute like this >> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >> >> >>>>> the call works fine. >> >>>>> >> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode >> >>>>> without forcing the codec string in the dial plan? >> >>>>> >> >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA >> and >> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM >> >>>>> >> >>>>> ---------------------------siptrace-------------------------------- >> >>>>> >> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >> >>>>> >> >>>>> >> >>>>> >> ------------------------------------------------------------------------ >> >>>>> SIP/2.0 406 Not Acceptable >> >>>>> Via: SIP/2.0/UDP >> >>>>> >> >>>>> >> >>>>> >> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >> >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> >>>>> From: "202" > >> >;tag=DFX0FUvr2vNcm >> >>>>> To: > >> >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> >>>>> CSeq: 87372504 INVITE >> >>>>> Content-Length: 0 >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> ------------------------------------------------------------------------ >> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >> >>>>> >> >>>>> >> >>>>> >> ------------------------------------------------------------------------ >> >>>>> ACK sip:303 at 192.168.1.5 >> SIP/2.0 >> >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >> >>>>> Max-Forwards: 68 >> >>>>> From: "202" > >> >;tag=DFX0FUvr2vNcm >> >>>>> To: > >> >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >> >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >> >>>>> CSeq: 87372504 ACK >> >>>>> Content-Length: 0 >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> ------------------------------------------------------------------------ >> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >> >>>>> sofia/internal/303 at 192.168.1.5 >> entering state >> [terminated][406] >> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >> >>>>> sofia/internal/303 at 192.168.1.5 >> [CS_CONSUME_MEDIA] >> >>>>> [SERVICE_NOT_IMPLEMENTED] >> >>>>> >> >>>>> Thanks >> >>>>> Rajil >> >>>>> >> >>>>> >> >>>>> >> _________________________________________________________________________ >> >>>>> Professional FreeSWITCH Consulting Services: >> >>>>> consulting at freeswitch.org >> >> >>>>> http://www.freeswitchsolutions.com >> >>>>> >> >>>>> Official FreeSWITCH Sites >> >>>>> http://www.freeswitch.org >> >>>>> http://confluence.freeswitch.org >> >>>>> http://www.cluecon.com >> >>>>> >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at 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/20160215/c2e6f5fa/attachment-0001.html From mike at jerris.com Mon Feb 15 18:35:31 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Feb 2016 10:35:31 -0500 Subject: [Freeswitch-users] bind_digit_action in a conference In-Reply-To: References: Message-ID: maybe if you entered a lua script and did those more advanced actions it could be done On Monday, February 15, 2016, Matt Broad wrote: > Hi, > > is it possible to bind a digit action and set the target as another member > of a conference? > > I would like to be able to press a digit and have another member execute > an extension where they enter some digits before returning back to the > conference. > > thanks > Matt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/7e769d6a/attachment.html From mitch.capper at gmail.com Mon Feb 15 18:51:02 2016 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 15 Feb 2016 07:51:02 -0800 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: I haven't had an issue with 14 or 15 with FS with the stock modules (and a few extras I use). ~mitch On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu wrote: > Hello, > > Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues > for several modules > > Silviu > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/ad30714b/attachment.html From matt at supportedbusiness.com Mon Feb 15 19:00:56 2016 From: matt at supportedbusiness.com (Matt Broad) Date: Mon, 15 Feb 2016 16:00:56 +0000 Subject: [Freeswitch-users] bind_digit_action in a conference In-Reply-To: References: Message-ID: Hi Michael, thanks for the response. I have tried this using javascript, but the digits are not collected. A quick example is I sit in a loop until the mod presses the relevant key, on the keypress I clone the session of the user who is to enter the digits and have that execute an extension. while (callConnected == true) { if(getDigits == true) { //set back to false to prevent this code block from being hit on the next loop getDigits = false; var _session = new session(uuid); _session.execute("execute_extension", "collectdigits XML Testfeatures"); } } If I set the bind_digit_action to the session that will enter the digits, it works fine but the issue is I would like the other member of the conference to instigate the request for digits _session.execute("bind_digit_action", "card_digits,6,exec:execute_extension,collectdigits XML Testfeatures,self,self"); thanks Matt On 15 February 2016 at 15:35, Michael Jerris wrote: > maybe if you entered a lua script and did those more advanced actions it > could be done > > > On Monday, February 15, 2016, Matt Broad > wrote: > >> Hi, >> >> is it possible to bind a digit action and set the target as another >> member of a conference? >> >> I would like to be able to press a digit and have another member execute >> an extension where they enter some digits before returning back to the >> conference. >> >> thanks >> Matt >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/362573bf/attachment.html From brian at freeswitch.org Mon Feb 15 19:03:01 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Feb 2016 10:03:01 -0600 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: For those interested https://www.gofundme.com/freeswitch_ubuntu We are working on Official packages, if you care to donate to the cause. /b On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper wrote: > I haven't had an issue with 14 or 15 with FS with the stock modules (and a > few extras I use). > > ~mitch > > On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu > wrote: > >> Hello, >> >> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues >> for several modules >> >> Silviu >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160215/c6edfccf/attachment-0001.html From colton.conor at gmail.com Mon Feb 15 19:04:13 2016 From: colton.conor at gmail.com (Colton Conor) Date: Mon, 15 Feb 2016 10:04:13 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> Message-ID: So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris wrote: > any device that even remotely follows sip > specs supports TCP. Most phones I have seen do. > > > On Sunday, February 14, 2016, Colton Conor wrote: > >> So is TCP the preferred method of doing SIP these days? I like TCP with >> endpoints as they always break through firewalls and we never seem to have >> in issue with TCP. However UDP is a headache. So if you have the choice why >> not do TCP? I realize some devices only support UDP, but the majority of >> SIP phones out there today do support TCP. >> >> Plus if you use TLS for encryption and security then you are already >> using TCP right? >> >> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: >> >>> This behavior changed a while ago. This was dictates by ever growing >>> SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with >>> fragmentation and everyone refuses to fully implement sip over tcp for some >>> reason even tho a ton of things support it and the RFCs require it >>> >>> Sent from my iPhone >>> >>> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: >>> > >>> > Thanks, after setting media_mix_inbound_outbound_codecs=true, >>> > transcoding happens automatically. I remember not setting this >>> > variable in other installations and transcoding used to work out of >>> > the box. Is media_mix_inbound_outbound_codecs=true default in >>> > Freeswitch? >>> > >>> >> On 14 February 2016 at 13:56, Russell Treleaven < >>> rtreleaven at bunnykick.ca> wrote: >>> >> fyi >>> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >>> >> >>> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat >>> wrote: >>> >>> >>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y >>> >>> >>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >>> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >>> >>> >>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli >> > >>> >>> wrote: >>> >>>> How you originate the call? Is a bridge? From which phone? >>> >>>> >>> >>>> Also, please pastebin the complete sip trace (from start of leg A >>> to end >>> >>>> of >>> >>>> both legs) and put here a link to pastebin >>> >>>> >>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha >>> scritto: >>> >>>>> >>> >>>>> Hello, >>> >>>>> >>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports >>> PCMU. >>> >>>>> When I call to this remote sip phone i get a 406 error that opus is >>> >>>>> not supported as shown by the sip trace below. However, if I force >>> the >>> >>>>> codec to absolute like this >>> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >>> >>>>> the call works fine. >>> >>>>> >>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode >>> >>>>> without forcing the codec string in the dial plan? >>> >>>>> >>> >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA >>> and >>> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM >>> >>>>> >>> >>>>> ---------------------------siptrace-------------------------------- >>> >>>>> >>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >>> >>>>> >>> >>>>> >>> >>>>> >>> ------------------------------------------------------------------------ >>> >>>>> SIP/2.0 406 Not Acceptable >>> >>>>> Via: SIP/2.0/UDP >>> >>>>> >>> >>>>> >>> >>>>> >>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >>> >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>> >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>> >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>> >>>>> CSeq: 87372504 INVITE >>> >>>>> Content-Length: 0 >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> ------------------------------------------------------------------------ >>> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >>> >>>>> >>> >>>>> >>> >>>>> >>> ------------------------------------------------------------------------ >>> >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 >>> >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >>> >>>>> Max-Forwards: 68 >>> >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>> >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>> >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>> >>>>> CSeq: 87372504 ACK >>> >>>>> Content-Length: 0 >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> ------------------------------------------------------------------------ >>> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >>> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >>> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >>> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >>> >>>>> [SERVICE_NOT_IMPLEMENTED] >>> >>>>> >>> >>>>> Thanks >>> >>>>> Rajil >>> >>>>> >>> >>>>> >>> >>>>> >>> _________________________________________________________________________ >>> >>>>> Professional FreeSWITCH Consulting Services: >>> >>>>> consulting at freeswitch.org >>> >>>>> http://www.freeswitchsolutions.com >>> >>>>> >>> >>>>> Official FreeSWITCH Sites >>> >>>>> http://www.freeswitch.org >>> >>>>> http://confluence.freeswitch.org >>> >>>>> http://www.cluecon.com >>> >>>>> >>> >>>>> FreeSWITCH-users mailing list >>> >>>>> FreeSWITCH-users at 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/20160215/823e3645/attachment-0001.html From mike at jerris.com Mon Feb 15 19:08:52 2016 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Feb 2016 11:08:52 -0500 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> Message-ID: <8402F14B-6417-4A62-98BF-C8C4CA07D425@jerris.com> It is heavier but I think that otherwise is superior. > On Feb 15, 2016, at 11:04 AM, Colton Conor wrote: > > So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris > wrote: > any device that even remotely follows sip > specs supports TCP. Most phones I have seen do. > > > On Sunday, February 14, 2016, Colton Conor > wrote: > So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. > > Plus if you use TLS for encryption and security then you are already using TCP right? > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice > wrote: > This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it > > Sent from my iPhone > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat > wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven > wrote: > >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) > >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > > >>> wrote: > >>>> How you originate the call? Is a bridge? From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A to end > >>>> of > >>>> both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" > ha scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus is > >>>>> not supported as shown by the sip trace below. However, if I force the > >>>>> codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 <> > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace-------------------------------- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" >;tag=DFX0FUvr2vNcm > >>>>> To: >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 <> SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" >;tag=DFX0FUvr2vNcm > >>>>> To: >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 <> entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 <> [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org <> > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/e5c2a565/attachment-0001.html From mandra at gmail.com Mon Feb 15 19:16:36 2016 From: mandra at gmail.com (Chris Mandra) Date: Mon, 15 Feb 2016 11:16:36 -0500 Subject: [Freeswitch-users] bridging out and custom signaling question Message-ID: Hi guys - I hope things are good where you are. I have a question: What is the easiest way to bridge a call out of freeswitch not using sip but using a custom signaling protocol that may be a restful protocol? Thank you ! chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/af63da0f/attachment.html From krice at freeswitch.org Mon Feb 15 19:25:11 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Feb 2016 10:25:11 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> Message-ID: <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> The problem isn?t necessarily the devices, but there is also the carriers? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor Sent: Monday, February 15, 2016 10:04 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris > wrote: any device that even remotely follows sip specs supports TCP. Most phones I have seen do. On Sunday, February 14, 2016, Colton Conor > wrote: So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. Plus if you use TLS for encryption and security then you are already using TCP right? On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice > wrote: This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it Sent from my iPhone > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat > wrote: > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > transcoding happens automatically. I remember not setting this > variable in other installations and transcoding used to work out of > the box. Is media_mix_inbound_outbound_codecs=true default in > Freeswitch? > >> On 14 February 2016 at 13:56, Russell Treleaven > wrote: >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: >>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y >>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> wrote: >>>> How you originate the call? Is a bridge? From which phone? >>>> >>>> Also, please pastebin the complete sip trace (from start of leg A to end >>>> of >>>> both legs) and put here a link to pastebin >>>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" > ha scritto: >>>>> >>>>> Hello, >>>>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. >>>>> When I call to this remote sip phone i get a 406 error that opus is >>>>> not supported as shown by the sip trace below. However, if I force the >>>>> codec to absolute like this >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >>>>> the call works fine. >>>>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode >>>>> without forcing the codec string in the dial plan? >>>>> >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and >>>>> outbound_codec_prefs=PCMU,PCMA,GSM >>>>> >>>>> ---------------------------siptrace-------------------------------- >>>>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 406 Not Acceptable >>>>> Via: SIP/2.0/UDP >>>>> >>>>> >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> CSeq: 87372504 INVITE >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >>>>> Max-Forwards: 68 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> CSeq: 87372504 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >>>>> [SERVICE_NOT_IMPLEMENTED] >>>>> >>>>> Thanks >>>>> Rajil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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/20160215/6870340f/attachment-0001.html From colton.conor at gmail.com Mon Feb 15 19:29:40 2016 From: colton.conor at gmail.com (Colton Conor) Date: Mon, 15 Feb 2016 10:29:40 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> Message-ID: True, But freeswitch talking to the carriers is almost always UPD. However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too :) On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice wrote: > The problem isn?t necessarily the devices, but there is also the carriers? > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colton Conor > *Sent:* Monday, February 15, 2016 10:04 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch doesnt transcode > > > > So if the device supports TCP, is there any reason not to use TCP. AKA is > there any reason to keep on using UDP. TCP seems superior. > > > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris wrote: > > any device that even remotely follows sip > > specs supports TCP. Most phones I have seen do. > > > > On Sunday, February 14, 2016, Colton Conor wrote: > > So is TCP the preferred method of doing SIP these days? I like TCP with > endpoints as they always break through firewalls and we never seem to have > in issue with TCP. However UDP is a headache. So if you have the choice why > not do TCP? I realize some devices only support UDP, but the majority of > SIP phones out there today do support TCP. > > > > Plus if you use TLS for encryption and security then you are already using > TCP right? > > > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: > > This behavior changed a while ago. This was dictates by ever growing SDPs > and exceeding MTUs causing udp fragmentation. Udp does not deal with > fragmentation and everyone refuses to fully implement sip over tcp for some > reason even tho a ton of things support it and the RFCs require it > > Sent from my iPhone > > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven < > rtreleaven at bunnykick.ca> wrote: > >> fyi > https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) > >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> wrote: > >>>> How you originate the call? Is a bridge? >From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A to > end > >>>> of > >>>> both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha > scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus is > >>>>> not supported as shown by the sip trace below. However, if I force > the > >>>>> codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA > and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace-------------------------------- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> > _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/8fd016be/attachment-0001.html From krice at freeswitch.org Mon Feb 15 19:35:05 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Feb 2016 10:35:05 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> Message-ID: <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> The problem still exists for expanding SDPs? using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place? so the decision was made to prevent those problems we?ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped? So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we?re stuck with that legacy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor Sent: Monday, February 15, 2016 10:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode True, But freeswitch talking to the carriers is almost always UPD. However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too :) On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice > wrote: The problem isn?t necessarily the devices, but there is also the carriers? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Colton Conor Sent: Monday, February 15, 2016 10:04 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris > wrote: any device that even remotely follows sip specs supports TCP. Most phones I have seen do. On Sunday, February 14, 2016, Colton Conor > wrote: So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. Plus if you use TLS for encryption and security then you are already using TCP right? On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice > wrote: This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it Sent from my iPhone > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat > wrote: > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > transcoding happens automatically. I remember not setting this > variable in other installations and transcoding used to work out of > the box. Is media_mix_inbound_outbound_codecs=true default in > Freeswitch? > >> On 14 February 2016 at 13:56, Russell Treleaven > wrote: >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: >>> >>> The siptrace is at http://pastebin.com/xiGqtj1Y >>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> wrote: >>>> How you originate the call? Is a bridge? >From which phone? >>>> >>>> Also, please pastebin the complete sip trace (from start of leg A to end >>>> of >>>> both legs) and put here a link to pastebin >>>> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" > ha scritto: >>>>> >>>>> Hello, >>>>> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. >>>>> When I call to this remote sip phone i get a 406 error that opus is >>>>> not supported as shown by the sip trace below. However, if I force the >>>>> codec to absolute like this >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 >>>>> the call works fine. >>>>> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode >>>>> without forcing the codec string in the dial plan? >>>>> >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and >>>>> outbound_codec_prefs=PCMU,PCMA,GSM >>>>> >>>>> ---------------------------siptrace-------------------------------- >>>>> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 406 Not Acceptable >>>>> Via: SIP/2.0/UDP >>>>> >>>>> >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> CSeq: 87372504 INVITE >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej >>>>> Max-Forwards: 68 >>>>> From: "202" ;tag=DFX0FUvr2vNcm >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 >>>>> CSeq: 87372504 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] >>>>> [SERVICE_NOT_IMPLEMENTED] >>>>> >>>>> Thanks >>>>> Rajil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160215/feb69bc4/attachment-0001.html From jalsot at gmail.com Mon Feb 15 20:39:06 2016 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Mon, 15 Feb 2016 18:39:06 +0100 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: Are you going to support upcoming 16.04 LTS too? Jalsot On Mon, Feb 15, 2016 at 5:03 PM, Brian West wrote: > For those interested https://www.gofundme.com/freeswitch_ubuntu > > We are working on Official packages, if you care to donate to the cause. > > /b > > On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper > wrote: > >> I haven't had an issue with 14 or 15 with FS with the stock modules (and >> a few extras I use). >> >> ~mitch >> >> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu >> wrote: >> >>> Hello, >>> >>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues >>> for several modules >>> >>> Silviu >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160215/8e9bdf11/attachment.html From brian at freeswitch.org Mon Feb 15 20:54:33 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Feb 2016 11:54:33 -0600 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: I don't see why not, it may require more testing and some work.... On Mon, Feb 15, 2016 at 11:39 AM, Tamas Jalsovszky wrote: > Are you going to support upcoming 16.04 LTS too? > > Jalsot > > On Mon, Feb 15, 2016 at 5:03 PM, Brian West wrote: > >> For those interested https://www.gofundme.com/freeswitch_ubuntu >> >> We are working on Official packages, if you care to donate to the cause. >> >> /b >> >> On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper >> wrote: >> >>> I haven't had an issue with 14 or 15 with FS with the stock modules (and >>> a few extras I use). >>> >>> ~mitch >>> >>> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu >>> wrote: >>> >>>> Hello, >>>> >>>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation >>>> issues for several modules >>>> >>>> Silviu >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 > -- *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/20160215/6aae39df/attachment-0001.html From bobjectsfreeswitch at gmail.com Mon Feb 15 21:27:26 2016 From: bobjectsfreeswitch at gmail.com (Bob Hartwig) Date: Mon, 15 Feb 2016 12:27:26 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven wrote: > fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > > That is a really lucid and complete description of CODEC negotiation in FS. Thanks for the link, and thanks to the original author for writing it. Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/da8c4730/attachment.html From lconroy at insensate.co.uk Mon Feb 15 21:44:38 2016 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Mon, 15 Feb 2016 18:44:38 +0000 Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) In-Reply-To: <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> Message-ID: Hi Ken, Conor, folks, Along time ago in a land far away ... the [very] early history of SIP was tied up with a bunch of other multimedia SxP things. The big driver at that time was being able to do conferencing and media distribution -- after all, voice calls could be done with H.323 et al. The SDP (and SIP main part headers) were quite simple and small [**]. This was a message based scheme, and UDP is a messaging transport, as opposed to TCP which is a stream transport. IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed simpler. TCP required maintaining transport session state in gateways, and the stacks in those gateways were primitive, to say the least. Despite that, folk pushing for TCP to be mandatory were told that it was considered 2nd class and should not be mandatory to implement); that was in '97 as I recall. Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to their will than H.323/224/..., and the number of headers grew like topsy, the complexity of the maintained state just kept on building, and we ran into fragment problems. Quick fix was header compression, but that ran into company-political issues in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed to be per week. THEN there was a drift away from UDP and towards TCP for purely practical reasons, and TCP became mandatory to implement (but NOT, of course mandatory to use, as there were any number of bits of kit out there that didn't have support for it :). Long story, but in short -- with the continued introduction of bloat (e.g., IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits. That shouldn't be a problem but is because frags are not dealt with well by end systems (as customer router/end system IP stacks tend to be nasty brutish and short on development). SO ... TCP has advantages (as long as your system can handle many parallel TCP sessions), is marginally slower on initial set up, but doesn't have to maintain the t30 et al timer stuff. From memory, getting the TCP stack tweaked for ultra-high load systems was a pain and led to obscure behaviour, but available TCP stacks seem generally better now. UDP was simpler to map to message based systems at gateways, didn't have to use good IP stacks as you were rolling your own logic, but given the lard that is SIP/SDP now, that's the least of your coding worries. For carriers, I understand why they have a reflex against maintaining state, and they're using kit that is "mature". It's hard to justify replacing kit that's familiar, has a management UI your staff know, and had its costs amortised away years ago; VoIP is not a high profit service so the bean counters WILL ask. => TCP may be 'better', but UDP is in kit that isn't going away soon. all the best, Lawrence **: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at Columbia University a post-grad course on IP comms during which he gave "implement a SIP-based voice call system" as a [two week] homework exercise, followed by interops between the clients. It had to be simple (and he was a "hard task master" [or words to that effect]; he knew that UDP forced all the timer logic to be coded as well). On 15 Feb 2016, at 16:35, Ken Rice wrote: > The problem still exists for expanding SDPs? using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place? so the decision was made to prevent those problems we?ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped? > > So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we?re stuck with that legacy > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor > Sent: Monday, February 15, 2016 10:30 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > True, > > But freeswitch talking to the carriers is almost always UPD. > > However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too :) > > On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice wrote: > The problem isn?t necessarily the devices, but there is also the carriers? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colton Conor > Sent: Monday, February 15, 2016 10:04 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris wrote: > any device that even remotely follows sip > specs supports TCP. Most phones I have seen do. > > > On Sunday, February 14, 2016, Colton Conor wrote: > So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. > > Plus if you use TLS for encryption and security then you are already using TCP right? > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: > This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it > > Sent from my iPhone > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven wrote: > >> fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) > >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> wrote: > >>>> How you originate the call? Is a bridge? >From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A to end > >>>> of > >>>> both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus is > >>>>> not supported as shown by the sip trace below. However, if I force the > >>>>> codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5 > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace-------------------------------- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> _________________________________________________________________________ > >>>>> Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org > >>>>> http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 krice at freeswitch.org Mon Feb 15 21:53:12 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Feb 2016 12:53:12 -0600 Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> Message-ID: <113801d16822$1f4cf910$5de6eb30$@freeswitch.org> Lawrence, Well Said! There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync so maybe we'll start seeing more traction there... In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy handed... something simple like Verto provides a lower overhead (in the browser) and allows for push/pull eventing... wish we would see wider adoption of such things in the near to mid terms -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence Conroy Sent: Monday, February 15, 2016 12:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) Hi Ken, Conor, folks, Along time ago in a land far away ... the [very] early history of SIP was tied up with a bunch of other multimedia SxP things. The big driver at that time was being able to do conferencing and media distribution -- after all, voice calls could be done with H.323 et al. The SDP (and SIP main part headers) were quite simple and small [**]. This was a message based scheme, and UDP is a messaging transport, as opposed to TCP which is a stream transport. IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed simpler. TCP required maintaining transport session state in gateways, and the stacks in those gateways were primitive, to say the least. Despite that, folk pushing for TCP to be mandatory were told that it was considered 2nd class and should not be mandatory to implement); that was in '97 as I recall. Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to their will than H.323/224/..., and the number of headers grew like topsy, the complexity of the maintained state just kept on building, and we ran into fragment problems. Quick fix was header compression, but that ran into company-political issues in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed to be per week. THEN there was a drift away from UDP and towards TCP for purely practical reasons, and TCP became mandatory to implement (but NOT, of course mandatory to use, as there were any number of bits of kit out there that didn't have support for it :). Long story, but in short -- with the continued introduction of bloat (e.g., IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits. That shouldn't be a problem but is because frags are not dealt with well by end systems (as customer router/end system IP stacks tend to be nasty brutish and short on development). SO ... TCP has advantages (as long as your system can handle many parallel TCP sessions), is marginally slower on initial set up, but doesn't have to maintain the t30 et al timer stuff. From memory, getting the TCP stack tweaked for ultra-high load systems was a pain and led to obscure behaviour, but available TCP stacks seem generally better now. UDP was simpler to map to message based systems at gateways, didn't have to use good IP stacks as you were rolling your own logic, but given the lard that is SIP/SDP now, that's the least of your coding worries. For carriers, I understand why they have a reflex against maintaining state, and they're using kit that is "mature". It's hard to justify replacing kit that's familiar, has a management UI your staff know, and had its costs amortised away years ago; VoIP is not a high profit service so the bean counters WILL ask. => TCP may be 'better', but UDP is in kit that isn't going away soon. all the best, Lawrence **: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at Columbia University a post-grad course on IP comms during which he gave "implement a SIP-based voice call system" as a [two week] homework exercise, followed by interops between the clients. It had to be simple (and he was a "hard task master" [or words to that effect]; he knew that UDP forced all the timer logic to be coded as well). On 15 Feb 2016, at 16:35, Ken Rice wrote: > The problem still exists for expanding SDPs. using TCP to the > user/device then trying to send the same thing out to the carrier over > UDP is what was causing the problem in the first place. so the > decision was made to prevent those problems we'll only offer what the > device offers and not expand the number of codecs even further > increasing the already bloated SDPs to the point where they fragment > over UDP and get dropped. > > So is TCP better for some things, yes it is, however, the lack of > market wide support for it with carriers makes it a pain in the ass > even tho the RFCs specifically say you MUST support both UDP and TCP > for SIP, but certain VoIP softwares out there only implemented UDP > many years ago and now we're stuck with that legacy > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Colton Conor > Sent: Monday, February 15, 2016 10:30 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > True, > > But freeswitch talking to the carriers is almost always UPD. > > However, freeswitch talking to the clients I would say TCP would be > idea. So its almost like freeswitch is trancoding from TCP to UDP too > :) > > On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice wrote: > The problem isn't necessarily the devices, but there is also the > carriers. > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Colton Conor > Sent: Monday, February 15, 2016 10:04 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris wrote: > any device that even remotely follows sip specs supports TCP. Most > phones I have seen do. > > > On Sunday, February 14, 2016, Colton Conor wrote: > So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. > > Plus if you use TLS for encryption and security then you are already using TCP right? > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: > This behavior changed a while ago. This was dictates by ever growing > SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal > with fragmentation and everyone refuses to fully implement sip over > tcp for some reason even tho a ton of things support it and the RFCs > require it > > Sent from my iPhone > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven wrote: > >> fyi > >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat > >> ion > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS > >>> codec) to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> > >>> wrote: > >>>> How you originate the call? Is a bridge? >From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A > >>>> to end of both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus > >>>>> is not supported as shown by the sip trace below. However, if I > >>>>> force the codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1. > >>>>> 5 > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as > >>>>> global_codec_prefs=OPUS,PCMU,PCMA and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace----------------------------- > >>>>> --- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ---------------------------------------------------------------- > >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at > >>>>> 08:02:16.368591: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ---------------------------------------------------------------- > >>>>> -------- > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> ________________________________________________________________ > >>>>> _________ Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit > >>>>> ch-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/freeswitc > >>>> h-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-u > > sers > > 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-use > rs > 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-use > rs > 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-use > rs > 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-use > rs > 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 Mon Feb 15 22:26:30 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Feb 2016 13:26:30 -0600 Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) In-Reply-To: <113801d16822$1f4cf910$5de6eb30$@freeswitch.org> References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> <113801d16822$1f4cf910$5de6eb30$@freeswitch.org> Message-ID: With SDP getting bigger and bigger with presence packates and large invites with many codecs or video and WebRTC TCP will become mandatory. The spec on when to use TCP is very arcane. Use UDP first unless the packet is > MTU, change to TCP. If the TCP times out (1 to 10 min) retry UDP anyway. With some fun mixed in like you MUST be under the MTU and you also MUST support packets over udp up to 64kb. Implementing that used to cause communications with asterisk to take forever because they only did UDP so bigger SDP packets would timeout on TCP first and everyone called it a bug. If you anticipate using presence or really big packets use TCP. If you use WebRTC its already TCP. On Mon, Feb 15, 2016 at 12:53 PM, Ken Rice wrote: > Lawrence, > > Well Said! > > There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync > so maybe we'll start seeing more traction there... > > In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy > handed... something simple like Verto provides a lower overhead (in the > browser) and allows for push/pull eventing... wish we would see wider > adoption of such things in the near to mid terms > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Lawrence > Conroy > Sent: Monday, February 15, 2016 12:45 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt > transcode) > > Hi Ken, Conor, folks, > Along time ago in a land far away ... the [very] early history of SIP was > tied up with a bunch of other multimedia SxP things. > The big driver at that time was being able to do conferencing and media > distribution -- after all, voice calls could be done with H.323 et al. The > SDP (and SIP main part headers) were quite simple and small [**]. This was > a > message based scheme, and UDP is a messaging transport, as opposed to TCP > which is a stream transport. > > IIRC, mapping from PSTN schemes (again, message-based systems) to UDP > seemed > simpler. > TCP required maintaining transport session state in gateways, and the > stacks > in those gateways were primitive, to say the least. > > Despite that, folk pushing for TCP to be mandatory were told that it was > considered 2nd class and should not be mandatory to implement); that was in > '97 as I recall. > > Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to > their will than H.323/224/..., and the number of headers grew like topsy, > the complexity of the maintained state just kept on building, and we ran > into fragment problems. > Quick fix was header compression, but that ran into company-political > issues > in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed > to be per week. THEN there was a drift away from UDP and towards TCP for > purely practical reasons, and TCP became mandatory to implement (but NOT, > of > course mandatory to use, as there were any number of bits of kit out there > that didn't have support for it :). > > Long story, but in short -- with the continued introduction of bloat (e.g., > IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits. > That shouldn't be a problem but is because frags are not dealt with well by > end systems (as customer router/end system IP stacks tend to be nasty > brutish and short on development). > > SO ... TCP has advantages (as long as your system can handle many parallel > TCP sessions), is marginally slower on initial set up, but doesn't have to > maintain the t30 et al timer stuff. From memory, getting the TCP stack > tweaked for ultra-high load systems was a pain and led to obscure > behaviour, > but available TCP stacks seem generally better now. > UDP was simpler to map to message based systems at gateways, didn't have to > use good IP stacks as you were rolling your own logic, but given the lard > that is SIP/SDP now, that's the least of your coding worries. > > For carriers, I understand why they have a reflex against maintaining > state, > and they're using kit that is "mature". It's hard to justify replacing kit > that's familiar, has a management UI your staff know, and had its costs > amortised away years ago; VoIP is not a high profit service so the bean > counters WILL ask. > > => TCP may be 'better', but UDP is in kit that isn't going away soon. > > all the best, > Lawrence > > > **: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at > Columbia University a post-grad course on IP comms during which he gave > "implement a SIP-based voice call system" as a [two week] homework > exercise, > followed by interops between the clients. It had to be simple (and he was a > "hard task master" [or words to that effect]; he knew that UDP forced all > the timer logic to be coded as well). > > On 15 Feb 2016, at 16:35, Ken Rice wrote: > > The problem still exists for expanding SDPs. using TCP to the > > user/device then trying to send the same thing out to the carrier over > > UDP is what was causing the problem in the first place. so the > > decision was made to prevent those problems we'll only offer what the > > device offers and not expand the number of codecs even further > > increasing the already bloated SDPs to the point where they fragment > > over UDP and get dropped. > > > > So is TCP better for some things, yes it is, however, the lack of > > market wide support for it with carriers makes it a pain in the ass > > even tho the RFCs specifically say you MUST support both UDP and TCP > > for SIP, but certain VoIP softwares out there only implemented UDP > > many years ago and now we're stuck with that legacy > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Colton Conor > > Sent: Monday, February 15, 2016 10:30 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > > > True, > > > > But freeswitch talking to the carriers is almost always UPD. > > > > However, freeswitch talking to the clients I would say TCP would be > > idea. So its almost like freeswitch is trancoding from TCP to UDP too > > :) > > > > On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice wrote: > > The problem isn't necessarily the devices, but there is also the > > carriers. > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Colton Conor > > Sent: Monday, February 15, 2016 10:04 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > > > So if the device supports TCP, is there any reason not to use TCP. AKA is > there any reason to keep on using UDP. TCP seems superior. > > > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris wrote: > > any device that even remotely follows sip specs supports TCP. Most > > phones I have seen do. > > > > > > On Sunday, February 14, 2016, Colton Conor > wrote: > > So is TCP the preferred method of doing SIP these days? I like TCP with > endpoints as they always break through firewalls and we never seem to have > in issue with TCP. However UDP is a headache. So if you have the choice why > not do TCP? I realize some devices only support UDP, but the majority of > SIP > phones out there today do support TCP. > > > > Plus if you use TLS for encryption and security then you are already > using > TCP right? > > > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice wrote: > > This behavior changed a while ago. This was dictates by ever growing > > SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal > > with fragmentation and everyone refuses to fully implement sip over > > tcp for some reason even tho a ton of things support it and the RFCs > > require it > > > > Sent from my iPhone > > > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat wrote: > > > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > > transcoding happens automatically. I remember not setting this > > > variable in other installations and transcoding used to work out of > > > the box. Is media_mix_inbound_outbound_codecs=true default in > > > Freeswitch? > > > > > >> On 14 February 2016 at 13:56, Russell Treleaven > wrote: > > >> fyi > > >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat > > >> ion > > >> > > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: > > >>> > > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > > >>> > > >>> The call is being made from 303 (Android/CSipsimple with OPUS > > >>> codec) to 208 (pjsua test client with PCMU codec). The error is on > line 545. > > >>> > > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > > >>> > > >>> wrote: > > >>>> How you originate the call? Is a bridge? >From which phone? > > >>>> > > >>>> Also, please pastebin the complete sip trace (from start of leg A > > >>>> to end of both legs) and put here a link to pastebin > > >>>> > > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" ha > scritto: > > >>>>> > > >>>>> Hello, > > >>>>> > > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports > PCMU. > > >>>>> When I call to this remote sip phone i get a 406 error that opus > > >>>>> is not supported as shown by the sip trace below. However, if I > > >>>>> force the codec to absolute like this > > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1. > > >>>>> 5 > > >>>>> the call works fine. > > >>>>> > > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > > >>>>> without forcing the codec string in the dial plan? > > >>>>> > > >>>>> The codec preferences is set as > > >>>>> global_codec_prefs=OPUS,PCMU,PCMA and > > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > > >>>>> > > >>>>> ---------------------------siptrace----------------------------- > > >>>>> --- > > >>>>> > > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > > >>>>> > > >>>>> > > >>>>> > ------------------------------------------------------------------------ > > >>>>> SIP/2.0 406 Not Acceptable > > >>>>> Via: SIP/2.0/UDP > > >>>>> > > >>>>> > > >>>>> > 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > > >>>>> CSeq: 87372504 INVITE > > >>>>> Content-Length: 0 > > >>>>> > > >>>>> > > >>>>> > > >>>>> ---------------------------------------------------------------- > > >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at > > >>>>> 08:02:16.368591: > > >>>>> > > >>>>> > > >>>>> > ------------------------------------------------------------------------ > > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > > >>>>> Max-Forwards: 68 > > >>>>> From: "202" ;tag=DFX0FUvr2vNcm > > >>>>> To: ;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > > >>>>> CSeq: 87372504 ACK > > >>>>> Content-Length: 0 > > >>>>> > > >>>>> > > >>>>> > > >>>>> ---------------------------------------------------------------- > > >>>>> -------- > > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > > >>>>> [SERVICE_NOT_IMPLEMENTED] > > >>>>> > > >>>>> Thanks > > >>>>> Rajil > > >>>>> > > >>>>> > > >>>>> ________________________________________________________________ > > >>>>> _________ Professional FreeSWITCH Consulting Services: > > >>>>> consulting at freeswitch.org > > >>>>> http://www.freeswitchsolutions.com > > >>>>> > > >>>>> Official FreeSWITCH Sites > > >>>>> http://www.freeswitch.org > > >>>>> http://confluence.freeswitch.org http://www.cluecon.com > > >>>>> > > >>>>> FreeSWITCH-users mailing list > > >>>>> FreeSWITCH-users at lists.freeswitch.org > > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>>>> > > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit > > >>>>> ch-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/freeswitc > > >>>> h-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-u > > > sers > > > 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-use > > rs > > 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-use > > rs > > 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-use > > rs > > 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-use > > rs > > 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 > -- 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/20160215/58c9fe86/attachment-0001.html From brian at freeswitch.org Mon Feb 15 22:35:11 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Feb 2016 13:35:11 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: google freeswitch behavior change we did send out warnings, change logged it, WIR'ed the change too. ;) Thanks, On Mon, Feb 15, 2016 at 12:27 PM, Bob Hartwig wrote: > On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven < > rtreleaven at bunnykick.ca> wrote: > >> fyi >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >> > That is a really lucid and complete description of CODEC negotiation in > FS. Thanks for the link, and thanks to the original author for writing it. > > 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 > -- *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/20160215/c2954a61/attachment.html From krice at freeswitch.org Mon Feb 15 22:48:31 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Feb 2016 13:48:31 -0600 Subject: [Freeswitch-users] Freeswitch doesnt transcode In-Reply-To: References: Message-ID: <118901d16829$d9803160$8c809420$@freeswitch.org> What? People are supposed to pay attention to announcements that the FreeSWITCH Dev Team makes?!@ what is this tom foolery? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, February 15, 2016 1:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode google freeswitch behavior change we did send out warnings, change logged it, WIR'ed the change too. ;) Thanks, On Mon, Feb 15, 2016 at 12:27 PM, Bob Hartwig > wrote: On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven > wrote: fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation That is a really lucid and complete description of CODEC negotiation in FS. Thanks for the link, and thanks to the original author for writing it. 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 -- 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/20160215/86d29fe7/attachment.html From italo at freeswitch.org Mon Feb 15 22:57:29 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 15 Feb 2016 16:57:29 -0300 Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: <1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com> References: <1597675782.6819.1455527545027.JavaMail.root@mailserver.edistar.com> Message-ID: It's probably due to the way you're dialing to your user, I can see in your log this bridge command: bridge(sofia/internal/4750%myserver) Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro wrote: > Thank you Italo. > > This is my log > > https://pastebin.freeswitch.org/24555 > > and this is the id of my user spy call > > 94bae5a0-dd2c-4d0e-99fd-8ef81e753227 > > Stefano > > ------------------------------ > *Da: *"?talo Rossi" > *A: *"FreeSWITCH Users Help" > *Inviato: *Gioved?, 11 febbraio 2016 18:52:46 > *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy > > > 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 < > stefano.favaro at edistar.com> 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/e466e3af/attachment-0001.html From antoine.durant29 at yahoo.fr Mon Feb 15 23:43:34 2016 From: antoine.durant29 at yahoo.fr (Antoine Durant) Date: Mon, 15 Feb 2016 20:43:34 +0000 (UTC) Subject: [Freeswitch-users] multiple internal ip References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com> Message-ID: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> Hi, I would like to have more internal IP addresses to do some "Multiple Companies". The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to the same provider sip. Is it possible to have an internal IP for each company: company-a.org => 192.168.1.1 company-b.org => 192.168.1.2 company-c.org => 192.168.1.3 company-X.org => 192.168.1.X How to use FreeSwitch this way? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/a3341a6c/attachment.html From brian at freeswitch.org Mon Feb 15 23:47:45 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Feb 2016 14:47:45 -0600 Subject: [Freeswitch-users] multiple internal ip In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com> <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> Message-ID: Why not use the DNS name? Only one IP is needed then. On Mon, Feb 15, 2016 at 2:43 PM, Antoine Durant wrote: > Hi, > > I would like to have more internal IP addresses to do some "Multiple > Companies". > > The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to > the same provider sip. > > Is it possible to have an internal IP for each company: > company-a.org => 192.168.1.1 > company-b.org => 192.168.1.2 > company-c.org => 192.168.1.3 > company-X.org => 192.168.1.X > > How to use FreeSwitch this way? > > 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 > -- *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/20160215/f54f9b2a/attachment.html From mgg at giagnocavo.net Tue Feb 16 00:55:53 2016 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Mon, 15 Feb 2016 21:55:53 +0000 Subject: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) In-Reply-To: References: <87D80ECA-87A9-4824-B9E7-4A1EE91AB4DC@freeswitch.org> <109501d1680d$723c40f0$56b4c2d0$@freeswitch.org> <109f01d1680e$d3baf190$7b30d4b0$@freeswitch.org> <113801d16822$1f4cf910$5de6eb30$@freeswitch.org> Message-ID: And that?s the reason as to why Lync is TCP-only. The commonly large payloads would always be above the SIP MTU meaning they?d need to switch to TCP. Instead of dealing with the pain of doing both (and the joy of adding that to interop tests), it was easier to simply do TCP-only. And really, people should just be using TLS anyways (but UDP+IPSec seems more common). Also FWIW, it?s easier to defend some attacks on TCP than UDP; at least it eliminates IP spoofing. (Enable syn cookies versus running a most-likely-buggy SIP parser on each message.) IP frag may not actually be an issue in practise. I looked at the signaling for a VoIP company with tens of thousands of end users. About 1% of all UDP SIP signaling was IP frag?d, though usually around 576 bytes. This had no noticeable impact on service (i.e., I didn?t see any related retransmits). But this is really just an anecdote. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, 15 February, 2016 13:27 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) With SDP getting bigger and bigger with presence packates and large invites with many codecs or video and WebRTC TCP will become mandatory. The spec on when to use TCP is very arcane. Use UDP first unless the packet is > MTU, change to TCP. If the TCP times out (1 to 10 min) retry UDP anyway. With some fun mixed in like you MUST be under the MTU and you also MUST support packets over udp up to 64kb. Implementing that used to cause communications with asterisk to take forever because they only did UDP so bigger SDP packets would timeout on TCP first and everyone called it a bug. If you anticipate using presence or really big packets use TCP. If you use WebRTC its already TCP. On Mon, Feb 15, 2016 at 12:53 PM, Ken Rice > wrote: Lawrence, Well Said! There is one upside., atleast Microsoft pushed the TCP stuff hard with Lync so maybe we'll start seeing more traction there... In reguards to the WebRTC stuff, imho SIP over WebRTC is a bit heavy handed... something simple like Verto provides a lower overhead (in the browser) and allows for push/pull eventing... wish we would see wider adoption of such things in the near to mid terms -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence Conroy Sent: Monday, February 15, 2016 12:45 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] TCP vs UDP (was Re: Freeswitch doesnt transcode) Hi Ken, Conor, folks, Along time ago in a land far away ... the [very] early history of SIP was tied up with a bunch of other multimedia SxP things. The big driver at that time was being able to do conferencing and media distribution -- after all, voice calls could be done with H.323 et al. The SDP (and SIP main part headers) were quite simple and small [**]. This was a message based scheme, and UDP is a messaging transport, as opposed to TCP which is a stream transport. IIRC, mapping from PSTN schemes (again, message-based systems) to UDP seemed simpler. TCP required maintaining transport session state in gateways, and the stacks in those gateways were primitive, to say the least. Despite that, folk pushing for TCP to be mandatory were told that it was considered 2nd class and should not be mandatory to implement); that was in '97 as I recall. Then (late 98 -> 2001) cable labs & 3GPP decided SIP was easier to bend to their will than H.323/224/..., and the number of headers grew like topsy, the complexity of the maintained state just kept on building, and we ran into fragment problems. Quick fix was header compression, but that ran into company-political issues in 3GPP and anyway couldn't keep up with the 5,000 new headers there seemed to be per week. THEN there was a drift away from UDP and towards TCP for purely practical reasons, and TCP became mandatory to implement (but NOT, of course mandatory to use, as there were any number of bits of kit out there that didn't have support for it :). Long story, but in short -- with the continued introduction of bloat (e.g., IMHO all the web RTC driven stuff) UDP is getting VERY tight on MTU limits. That shouldn't be a problem but is because frags are not dealt with well by end systems (as customer router/end system IP stacks tend to be nasty brutish and short on development). SO ... TCP has advantages (as long as your system can handle many parallel TCP sessions), is marginally slower on initial set up, but doesn't have to maintain the t30 et al timer stuff. From memory, getting the TCP stack tweaked for ultra-high load systems was a pain and led to obscure behaviour, but available TCP stacks seem generally better now. UDP was simpler to map to message based systems at gateways, didn't have to use good IP stacks as you were rolling your own logic, but given the lard that is SIP/SDP now, that's the least of your coding worries. For carriers, I understand why they have a reflex against maintaining state, and they're using kit that is "mature". It's hard to justify replacing kit that's familiar, has a management UI your staff know, and had its costs amortised away years ago; VoIP is not a high profit service so the bean counters WILL ask. => TCP may be 'better', but UDP is in kit that isn't going away soon. all the best, Lawrence **: Remember, at the time ('97-'98) Henning Schulzrinne was teaching at Columbia University a post-grad course on IP comms during which he gave "implement a SIP-based voice call system" as a [two week] homework exercise, followed by interops between the clients. It had to be simple (and he was a "hard task master" [or words to that effect]; he knew that UDP forced all the timer logic to be coded as well). On 15 Feb 2016, at 16:35, Ken Rice > wrote: > The problem still exists for expanding SDPs. using TCP to the > user/device then trying to send the same thing out to the carrier over > UDP is what was causing the problem in the first place. so the > decision was made to prevent those problems we'll only offer what the > device offers and not expand the number of codecs even further > increasing the already bloated SDPs to the point where they fragment > over UDP and get dropped. > > So is TCP better for some things, yes it is, however, the lack of > market wide support for it with carriers makes it a pain in the ass > even tho the RFCs specifically say you MUST support both UDP and TCP > for SIP, but certain VoIP softwares out there only implemented UDP > many years ago and now we're stuck with that legacy > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Colton Conor > Sent: Monday, February 15, 2016 10:30 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > True, > > But freeswitch talking to the carriers is almost always UPD. > > However, freeswitch talking to the clients I would say TCP would be > idea. So its almost like freeswitch is trancoding from TCP to UDP too > :) > > On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice > wrote: > The problem isn't necessarily the devices, but there is also the > carriers. > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Colton Conor > Sent: Monday, February 15, 2016 10:04 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode > > So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. > > On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris > wrote: > any device that even remotely follows sip specs supports TCP. Most > phones I have seen do. > > > On Sunday, February 14, 2016, Colton Conor > wrote: > So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP. > > Plus if you use TLS for encryption and security then you are already using TCP right? > > On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice > wrote: > This behavior changed a while ago. This was dictates by ever growing > SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal > with fragmentation and everyone refuses to fully implement sip over > tcp for some reason even tho a ton of things support it and the RFCs > require it > > Sent from my iPhone > > > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat > wrote: > > > > Thanks, after setting media_mix_inbound_outbound_codecs=true, > > transcoding happens automatically. I remember not setting this > > variable in other installations and transcoding used to work out of > > the box. Is media_mix_inbound_outbound_codecs=true default in > > Freeswitch? > > > >> On 14 February 2016 at 13:56, Russell Treleaven > wrote: > >> fyi > >> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiat > >> ion > >> > >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat > wrote: > >>> > >>> The siptrace is at http://pastebin.com/xiGqtj1Y > >>> > >>> The call is being made from 303 (Android/CSipsimple with OPUS > >>> codec) to 208 (pjsua test client with PCMU codec). The error is on line 545. > >>> > >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli > >>> > > >>> wrote: > >>>> How you originate the call? Is a bridge? >From which phone? > >>>> > >>>> Also, please pastebin the complete sip trace (from start of leg A > >>>> to end of both legs) and put here a link to pastebin > >>>> > >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" > ha scritto: > >>>>> > >>>>> Hello, > >>>>> > >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU. > >>>>> When I call to this remote sip phone i get a 406 error that opus > >>>>> is not supported as shown by the sip trace below. However, if I > >>>>> force the codec to absolute like this > >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1. > >>>>> 5 > >>>>> the call works fine. > >>>>> > >>>>> Is there anyway I can make FreeSWITCH to automatically transcode > >>>>> without forcing the codec string in the dial plan? > >>>>> > >>>>> The codec preferences is set as > >>>>> global_codec_prefs=OPUS,PCMU,PCMA and > >>>>> outbound_codec_prefs=PCMU,PCMA,GSM > >>>>> > >>>>> ---------------------------siptrace----------------------------- > >>>>> --- > >>>>> > >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> SIP/2.0 406 Not Acceptable > >>>>> Via: SIP/2.0/UDP > >>>>> > >>>>> > >>>>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> From: "202" >;tag=DFX0FUvr2vNcm > >>>>> To: >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> CSeq: 87372504 INVITE > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ---------------------------------------------------------------- > >>>>> -------- send 324 bytes to udp/[192.168.1.5]:5060 at > >>>>> 08:02:16.368591: > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ > >>>>> ACK sip:303 at 192.168.1.5 SIP/2.0 > >>>>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej > >>>>> Max-Forwards: 68 > >>>>> From: "202" >;tag=DFX0FUvr2vNcm > >>>>> To: >;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q > >>>>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 > >>>>> CSeq: 87372504 ACK > >>>>> Content-Length: 0 > >>>>> > >>>>> > >>>>> > >>>>> ---------------------------------------------------------------- > >>>>> -------- > >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel > >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406] > >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup > >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA] > >>>>> [SERVICE_NOT_IMPLEMENTED] > >>>>> > >>>>> Thanks > >>>>> Rajil > >>>>> > >>>>> > >>>>> ________________________________________________________________ > >>>>> _________ Professional FreeSWITCH Consulting Services: > >>>>> consulting at freeswitch.org > >>>>> http://www.freeswitchsolutions.com > >>>>> > >>>>> Official FreeSWITCH Sites > >>>>> http://www.freeswitch.org > >>>>> http://confluence.freeswitch.org http://www.cluecon.com > >>>>> > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit > >>>>> ch-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/freeswitc > >>>> h-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-u > > sers > > 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-use > rs > 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-use > rs > 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-use > rs > 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-use > rs > 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 -- 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/20160215/d152f891/attachment-0001.html From mgg at giagnocavo.net Tue Feb 16 01:01:52 2016 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Mon, 15 Feb 2016 22:01:52 +0000 Subject: [Freeswitch-users] High availability on different networks In-Reply-To: <1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch> References: <78429801-FCA4-4BA9-8EA6-64A1BD7ECA28@kavun.ch> <1382DEBE-2DB3-48AC-AE54-5AD9EE370709@kavun.ch> Message-ID: For outbound calls, you should be fine as long as your frontend SIP proxies can reach any backend (media/app/whatever) server. This is exactly what Via is for, right? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah Sent: Sunday, 14 February, 2016 10:18 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] High availability on different networks I love all your ideas, and thanks for sharing. The best option remains to test things out in real scenarios and see what happens. I?m already distributing my media traffic. Now I?d like to make the SIP part redundant. Just got some server resources in different locations. I?ll report back with my findings! E On Feb 12, 2016, at 10:09 AM, Andrew Cassidy > 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 [http://www.cassidywebservices.co.uk/static/emailsig.png] 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/20160215/772e18a2/attachment-0001.html From lists at telefaks.de Tue Feb 16 01:36:43 2016 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 15 Feb 2016 23:36:43 +0100 Subject: [Freeswitch-users] multiple internal ip In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com> <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> Message-ID: <56C252FB.6060808@telefaks.de> We have build a system with several IPs including multiple VPNs. For this purpose, we created a number of profiles, one for each IP and grouped them to contextes. This is possible, but you neeed to take care about your dialstrings when dialling to phones in distinct profiles. This may be more difficult with static xml configs, so with xml-curl it can be handled in a better way. As Brian commented, a multi tenant setup can be setup with distinct domains. But there may be of course other reasons for using distinct IPs. Bonne chance! Peter On 02/15/16 21:43, Antoine Durant wrote: > Hi, > > I would like to have more internal IP addresses to do some "Multiple > Companies". > > The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by > company) to the same provider sip. > > Is it possible to have an internal IP for each company: > company-a.org => 192.168.1.1 > company-b.org => 192.168.1.2 > company-c.org => 192.168.1.3 > company-X.org => 192.168.1.X > > How to use FreeSwitch this way? > > 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 -- 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/20160215/a142243e/attachment.html From ssinyagin at gmail.com Tue Feb 16 01:43:46 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 15 Feb 2016 23:43:46 +0100 Subject: [Freeswitch-users] multiple internal ip In-Reply-To: <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com> <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> Message-ID: each SIP profile binds to a unique combination of IP address and TCP+UDP port. So, you can define as many SIP profiles as you need, and route the inbound calls into different contexts as needed. But as other colleagues explained, distinctive DNS domains are usually sufficient for a multi-tenancy setup. I outlined some details in this article: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example On Mon, Feb 15, 2016 at 9:43 PM, Antoine Durant wrote: > Hi, > > I would like to have more internal IP addresses to do some "Multiple > Companies". > > The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to > the same provider sip. > > Is it possible to have an internal IP for each company: > company-a.org => 192.168.1.1 > company-b.org => 192.168.1.2 > company-c.org => 192.168.1.3 > company-X.org => 192.168.1.X > > How to use FreeSwitch this way? > > 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 From rajil.s at gmail.com Tue Feb 16 01:44:53 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Mon, 15 Feb 2016 16:44:53 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? Message-ID: Hello, I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102. My connections is as follows: ATA_A---->FS_A---->vpn---->FS_B---->ATA_B When user A is calling to B where should i specify jitter buffer amongst the following: 1. The bridge between Freeswitch instance FS_A and FS_B 2. The bridge between Freeswitch instance FS_B and ATA_B Thanks, Rajil From andretodd at verizon.net Tue Feb 16 02:03:45 2016 From: andretodd at verizon.net (Andre DeMattia) Date: Mon, 15 Feb 2016 18:03:45 -0500 Subject: [Freeswitch-users] SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA Message-ID: <061201d16845$1f679810$5e36c830$@verizon.net> Hi is there an application that will give me the same results as this event? switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA I see execute_on_ring and execute_on_media. Wondering if either of these are the same as switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA Thanks Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/7dea3dee/attachment.html From anthony.minessale at gmail.com Tue Feb 16 02:06:35 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Feb 2016 17:06:35 -0600 Subject: [Freeswitch-users] SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA In-Reply-To: <061201d16845$1f679810$5e36c830$@verizon.net> References: <061201d16845$1f679810$5e36c830$@verizon.net> Message-ID: execute_on_pre_answer On Mon, Feb 15, 2016 at 5:03 PM, Andre DeMattia wrote: > Hi is there an application that will give me the same results as this > event? switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA I see > execute_on_ring > and > execute_on_media. Wondering if either of these are the same as > switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA > > > > Thanks > > Andre > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/7ca5e969/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 16 02:10:11 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Feb 2016 17:10:11 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? In-Reply-To: References: Message-ID: The variable must be set on the leg before the channel engages media. you can test if its on with uuid_jitterbuffer debug:10 if you want it on both legs set the following global vars in vars.xml rtp_jitter_buffer_during_bridge=true jitterbuffer_msec=60 On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat wrote: > Hello, > > I have two FS instances connected to each other over VPN. The VPN > sometimes has poor connectivity and I was wondering whether > jitterbuffer can help to improve the call quality. I am using the OPUS > codec between the FS instances since it is supposed to behave better > in lower bandwidth. The ATA on both ends are Linksys SPA3102. > > My connections is as follows: > > ATA_A---->FS_A---->vpn---->FS_B---->ATA_B > > > When user A is calling to B where should i specify jitter buffer > amongst the following: > > 1. The bridge between Freeswitch instance FS_A and FS_B > data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/> > > 2. The bridge between Freeswitch instance FS_B and ATA_B > data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/> > > Thanks, > Rajil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/b19504e8/attachment.html From brian at freeswitch.org Tue Feb 16 02:48:19 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Feb 2016 17:48:19 -0600 Subject: [Freeswitch-users] multiple internal ip In-Reply-To: References: <857665560.7491164.1455569014252.JavaMail.yahoo.ref@mail.yahoo.com> <857665560.7491164.1455569014252.JavaMail.yahoo@mail.yahoo.com> Message-ID: You can if you want put multiple domains on a single profile... just sayin! :) On Mon, Feb 15, 2016 at 4:43 PM, Stanislav Sinyagin wrote: > each SIP profile binds to a unique combination of IP address and > TCP+UDP port. So, you can define as many SIP profiles as you need, and > route the inbound calls into different contexts as needed. > > But as other colleagues explained, distinctive DNS domains are usually > sufficient for a multi-tenancy setup. > > I outlined some details in this article: > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example > > > > > > On Mon, Feb 15, 2016 at 9:43 PM, Antoine Durant > wrote: > > Hi, > > > > I would like to have more internal IP addresses to do some "Multiple > > Companies". > > > > The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) > to > > the same provider sip. > > > > Is it possible to have an internal IP for each company: > > company-a.org => 192.168.1.1 > > company-b.org => 192.168.1.2 > > company-c.org => 192.168.1.3 > > company-X.org => 192.168.1.X > > > > How to use FreeSwitch this way? > > > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/6c07b5d6/attachment.html From rajil.s at gmail.com Tue Feb 16 02:54:54 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Mon, 15 Feb 2016 17:54:54 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? In-Reply-To: References: Message-ID: The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says, If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality. So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says? On 15 February 2016 at 17:10, Anthony Minessale wrote: > The variable must be set on the leg before the channel engages media. > you can test if its on with > > uuid_jitterbuffer debug:10 > > > if you want it on both legs set the following global vars in vars.xml > > rtp_jitter_buffer_during_bridge=true > > jitterbuffer_msec=60 > > > > > > > On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat wrote: > >> Hello, >> >> I have two FS instances connected to each other over VPN. The VPN >> sometimes has poor connectivity and I was wondering whether >> jitterbuffer can help to improve the call quality. I am using the OPUS >> codec between the FS instances since it is supposed to behave better >> in lower bandwidth. The ATA on both ends are Linksys SPA3102. >> >> My connections is as follows: >> >> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B >> >> >> When user A is calling to B where should i specify jitter buffer >> amongst the following: >> >> 1. The bridge between Freeswitch instance FS_A and FS_B >> > data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/> >> >> 2. The bridge between Freeswitch instance FS_B and ATA_B >> > data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/> >> >> Thanks, >> Rajil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/bf3ab70d/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 16 03:23:01 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Feb 2016 18:23:01 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? In-Reply-To: References: Message-ID: If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end. That is why its not defaulted to being possible but you asked how to do it. On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat wrote: > > The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says, > > If both sides of a bridge are RTP and both sides have a jb, its fairly > useless. In fact if anything, it can worsen call quality. > > So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says? > > > On 15 February 2016 at 17:10, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The variable must be set on the leg before the channel engages media. >> you can test if its on with >> >> uuid_jitterbuffer debug:10 >> >> >> if you want it on both legs set the following global vars in vars.xml >> >> rtp_jitter_buffer_during_bridge=true >> >> jitterbuffer_msec=60 >> >> >> >> >> >> >> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat >> wrote: >> >>> Hello, >>> >>> I have two FS instances connected to each other over VPN. The VPN >>> sometimes has poor connectivity and I was wondering whether >>> jitterbuffer can help to improve the call quality. I am using the OPUS >>> codec between the FS instances since it is supposed to behave better >>> in lower bandwidth. The ATA on both ends are Linksys SPA3102. >>> >>> My connections is as follows: >>> >>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B >>> >>> >>> When user A is calling to B where should i specify jitter buffer >>> amongst the following: >>> >>> 1. The bridge between Freeswitch instance FS_A and FS_B >>> >> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/> >>> >>> 2. The bridge between Freeswitch instance FS_B and ATA_B >>> >> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/> >>> >>> Thanks, >>> Rajil >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/9a608558/attachment.html From rajil.s at gmail.com Tue Feb 16 03:33:59 2016 From: rajil.s at gmail.com (Rajil Saraswat) Date: Mon, 15 Feb 2016 18:33:59 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? In-Reply-To: References: Message-ID: Ok, what about my specific use case where both endpoints ATA (Linksys spa3102) do have internal jitterbuffer. Should i still set the global variables as you suggested to improve the call quality over vpn? On 15 February 2016 at 18:23, Anthony Minessale wrote: > If one or the other end of the call does not have its own JB then putting > it on both sides can fix the stream for the other end. > That is why its not defaulted to being possible but you asked how to do it. > > > On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat wrote: > >> >> The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says, >> >> If both sides of a bridge are RTP and both sides have a jb, its fairly >> useless. In fact if anything, it can worsen call quality. >> >> So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says? >> >> >> On 15 February 2016 at 17:10, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The variable must be set on the leg before the channel engages media. >>> you can test if its on with >>> >>> uuid_jitterbuffer debug:10 >>> >>> >>> if you want it on both legs set the following global vars in vars.xml >>> >>> rtp_jitter_buffer_during_bridge=true >>> >>> jitterbuffer_msec=60 >>> >>> >>> >>> >>> >>> >>> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat >>> wrote: >>> >>>> Hello, >>>> >>>> I have two FS instances connected to each other over VPN. The VPN >>>> sometimes has poor connectivity and I was wondering whether >>>> jitterbuffer can help to improve the call quality. I am using the OPUS >>>> codec between the FS instances since it is supposed to behave better >>>> in lower bandwidth. The ATA on both ends are Linksys SPA3102. >>>> >>>> My connections is as follows: >>>> >>>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B >>>> >>>> >>>> When user A is calling to B where should i specify jitter buffer >>>> amongst the following: >>>> >>>> 1. The bridge between Freeswitch instance FS_A and FS_B >>>> >>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/> >>>> >>>> 2. The bridge between Freeswitch instance FS_B and ATA_B >>>> >>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/> >>>> >>>> Thanks, >>>> Rajil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/d3477aaa/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 16 03:46:01 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Feb 2016 18:46:01 -0600 Subject: [Freeswitch-users] Where to put Jitterbuffer? In-Reply-To: References: Message-ID: When you keep the codec the same across the whole call path then the jitterbuffer is not as important because FS will preserve the timestamps. If you are transcoding or recording the call, then the JB can help because you are creating new timestamps so the far end can no longer perceive the jitter. Ideally you should just try it. Also you should make sure your vpn is using udp or gprs not tcp You should probably try it with everything PCMU or PCMA as well with and without the JB. Trial and error is your friend. But without the vars I mentioned it will never use the JB during a bridge. On Mon, Feb 15, 2016 at 6:33 PM, Rajil Saraswat wrote: > Ok, what about my specific use case where both endpoints ATA (Linksys > spa3102) do have internal jitterbuffer. Should i still set the global > variables as you suggested to improve the call quality over vpn? > > On 15 February 2016 at 18:23, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> If one or the other end of the call does not have its own JB then putting >> it on both sides can fix the stream for the other end. >> That is why its not defaulted to being possible but you asked how to do >> it. >> >> >> On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat >> wrote: >> >>> >>> The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says, >>> >>> If both sides of a bridge are RTP and both sides have a jb, its fairly >>> useless. In fact if anything, it can worsen call quality. >>> >>> So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says? >>> >>> >>> On 15 February 2016 at 17:10, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The variable must be set on the leg before the channel engages media. >>>> you can test if its on with >>>> >>>> uuid_jitterbuffer debug:10 >>>> >>>> >>>> if you want it on both legs set the following global vars in vars.xml >>>> >>>> rtp_jitter_buffer_during_bridge=true >>>> >>>> jitterbuffer_msec=60 >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> I have two FS instances connected to each other over VPN. The VPN >>>>> sometimes has poor connectivity and I was wondering whether >>>>> jitterbuffer can help to improve the call quality. I am using the OPUS >>>>> codec between the FS instances since it is supposed to behave better >>>>> in lower bandwidth. The ATA on both ends are Linksys SPA3102. >>>>> >>>>> My connections is as follows: >>>>> >>>>> ATA_A---->FS_A---->vpn---->FS_B---->ATA_B >>>>> >>>>> >>>>> When user A is calling to B where should i specify jitter buffer >>>>> amongst the following: >>>>> >>>>> 1. The bridge between Freeswitch instance FS_A and FS_B >>>>> >>>> data="{jitterbuffer_msec=60}sofia/gateway/$1 at FS_B_IP"/> >>>>> >>>>> 2. The bridge between Freeswitch instance FS_B and ATA_B >>>>> >>>> data="{jitterbuffer_msec=60}sofia/internal/$1 at ATA_B_IP"/> >>>>> >>>>> Thanks, >>>>> Rajil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/ab136853/attachment.html From mandra at gmail.com Tue Feb 16 07:20:44 2016 From: mandra at gmail.com (Chris Mandra) Date: Mon, 15 Feb 2016 23:20:44 -0500 Subject: [Freeswitch-users] bridging out and custom signaling question In-Reply-To: References: Message-ID: Any ideas on this? On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra > wrote: > Hi guys - I hope things are good where you are. I have a question: > > What is the easiest way to bridge a call out of freeswitch not using sip > but using a custom signaling protocol that may be a restful protocol? > > Thank you ! > > chris > > -- mandra c:410.258.5281 -- mandra c:410.258.5281 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/d96b0731/attachment-0001.html From krice at freeswitch.org Tue Feb 16 07:55:34 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Feb 2016 22:55:34 -0600 Subject: [Freeswitch-users] bridging out and custom signaling question In-Reply-To: References: Message-ID: <8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org> You'll have to write (or have written) an endpoint driver for it... Sent from my iPhone > On Feb 15, 2016, at 10:20 PM, Chris Mandra wrote: > > Any ideas on this? >> On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra wrote: >> Hi guys - I hope things are good where you are. I have a question: >> >> What is the easiest way to bridge a call out of freeswitch not using sip but using a custom signaling protocol that may be a restful protocol? >> >> Thank you ! >> >> chris >> > > > > -- > mandra > c:410.258.5281 > > > -- > mandra > c:410.258.5281 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160215/d796d99e/attachment.html From gmaruzz at gmail.com Tue Feb 16 10:57:44 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 16 Feb 2016 08:57:44 +0100 Subject: [Freeswitch-users] bridging out and custom signaling question In-Reply-To: <8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org> References: <8DB037F6-0515-41A5-B265-7C3684B68433@freeswitch.org> Message-ID: You can have a look at mod-alsa (is very easy) and search into http://www.freeswitch.org/confluence for info about writing endpoints in C. Also, you can write at consulting at freeswitch.org for official, professional FreeSWITCH development and deployment. Il 16/Feb/2016 05:56, "Ken Rice" ha scritto: > You'll have to write (or have written) an endpoint driver for it... > > Sent from my iPhone > > On Feb 15, 2016, at 10:20 PM, Chris Mandra wrote: > > Any ideas on this? > On Mon, Feb 15, 2016 at 11:16 AM, Chris Mandra wrote: > >> Hi guys - I hope things are good where you are. I have a question: >> >> What is the easiest way to bridge a call out of freeswitch not using sip >> but using a custom signaling protocol that may be a restful protocol? >> >> Thank you ! >> >> chris >> >> > > > -- > mandra > c:410.258.5281 > > > -- > mandra > c:410.258.5281 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/b8745eb3/attachment.html From stefano.favaro at edistar.com Tue Feb 16 11:40:43 2016 From: stefano.favaro at edistar.com (Stefano Favaro) Date: Tue, 16 Feb 2016 09:40:43 +0100 (CET) Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: Message-ID: <775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com> Thanks, I've tried with but unfortunately I have the same problem. SF. https://pastebin.freeswitch.org/24556 ----- Messaggio originale ----- Da: "?talo Rossi" A: "FreeSWITCH Users Help" Inviato: Luned?, 15 febbraio 2016 20:57:29 Oggetto: Re: [Freeswitch-users] Problem with mod_spy It's probably due to the way you're dialing to your user, I can see in your log this bridge command: bridge(sofia/internal/4750%myserver) Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: Thank you Italo. This is my log https://pastebin.freeswitch.org/24555 and this is the id of my user spy call 94bae5a0-dd2c-4d0e-99fd-8ef81e753227 Stefano Da: "?talo Rossi" < italo at freeswitch.org > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Inviato: Gioved?, 11 febbraio 2016 18:52:46 Oggetto: Re: [Freeswitch-users] Problem with mod_spy 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 < stefano.favaro at edistar.com > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160216/ea18c3b0/attachment-0001.html From pskoul at gmail.com Tue Feb 16 14:35:38 2016 From: pskoul at gmail.com (Panagiotis Skoulikaritis) Date: Tue, 16 Feb 2016 13:35:38 +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: <56B7A25A.8090801@gmail.com> References: <56AE42E1.3080703@gmail.com> <3CBD15A5-819B-487B-9DE0-C8120DDCAC94@gmail.com> <3D075DBF-8FEF-40CE-9922-A7CFC462512E@gmail.com> <56B7A25A.8090801@gmail.com> Message-ID: <56C3098A.1040000@gmail.com> What possibly could cause the account code to not be written in the cdr? I have seen from traces that it is present on the sip mesage, Unfortunately the loging is disabled I will enable it to see if I can see anything. But from what I see is that it doesn't write the account code on the cdr when the outgoing leg for whatever reason is not made. any advice ? Regards Panagiotis On 7/2/2016 10:00 ??, Panagiotis Skoulikaritis wrote: > 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 italo at freeswitch.org Tue Feb 16 15:47:54 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 16 Feb 2016 09:47:54 -0300 Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: <775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com> References: <775731365.7256.1455612043715.JavaMail.root@mailserver.edistar.com> Message-ID: I mean your bridge string, not userspy, instead of sofia/internal/4750 at ... just use bridge(user/4750) to DIAL to that user. Your userspy args looks good On Tue, Feb 16, 2016 at 5:40 AM, Stefano Favaro wrote: > Thanks, > > I've tried with > > > > expression="^88(.*)$|^\*0(.*)$"> > > > > > > > but unfortunately I have the same problem. > > SF. > > https://pastebin.freeswitch.org/24556 > > > ------------------------------ > *Da: *"?talo Rossi" > *A: *"FreeSWITCH Users Help" > *Inviato: *Luned?, 15 febbraio 2016 20:57:29 > > *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy > > It's probably due to the way you're dialing to your user, I can see in > your log this bridge command: bridge(sofia/internal/4750%myserver) > > Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. > > On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro < > stefano.favaro at edistar.com> wrote: > >> Thank you Italo. >> >> This is my log >> >> https://pastebin.freeswitch.org/24555 >> >> and this is the id of my user spy call >> >> 94bae5a0-dd2c-4d0e-99fd-8ef81e753227 >> >> Stefano >> >> ------------------------------ >> *Da: *"?talo Rossi" >> *A: *"FreeSWITCH Users Help" >> *Inviato: *Gioved?, 11 febbraio 2016 18:52:46 >> *Oggetto: *Re: [Freeswitch-users] Problem with mod_spy >> >> >> 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 < >> stefano.favaro at edistar.com> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/063bf4be/attachment.html From royj at yandex.ru Tue Feb 16 15:58:37 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Tue, 16 Feb 2016 15:58:37 +0300 Subject: [Freeswitch-users] hiredis: realm must be defined Message-ID: <511661455627517@web2g.yandex.ru> Hi, all Can anybody clarify what mean 'hiredis: realm must be defined'? Where is the misconfiguration? I am trying to use hiredis backend for limits and call api like 'api limit_usage hiredis myrealm id', got api/response '-1' and FreeSWITCH says in log 'hiredis: realm must be defined'. Module mod_hiredis loaded and exists. For hash backend works fine. From aqsyounas at gmail.com Tue Feb 16 16:35:12 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 16 Feb 2016 18:35:12 +0500 Subject: [Freeswitch-users] 400 Bad To Header. Message-ID: Hi users, My freeswitch is replying with me 400 Bad To Header whenever It receives Bye containing this To Header. To: 1111#15122031234 ;tag=NKU2jrZvFaK0j But I wonder in the case of Invite containing, this header everything is fine. Is there something wrong with this header? (#?) Best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/558ec468/attachment.html From mitch.capper at gmail.com Tue Feb 16 17:22:44 2016 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 16 Feb 2016 06:22:44 -0800 Subject: [Freeswitch-users] freeswitch 1.6 on Ubuntu 14.04 LTS In-Reply-To: References: Message-ID: Official packages sound great! For those who do want to compile from source for now a good starter may be the dockerfile i use for ubuntu. I simplified version is attached to the ticket at: https://freeswitch.org/jira/browse/FS-7833 you don't need to use docker the commands in the file(its just text), after the "RUN " prefix are just shell commands so just copying and pasting them into an ubuntu box should work. I think that was from ubuntu 15 officially but i don't think I changed the packages much from 14 (if you run into any issues let me know and I can find my old one). ~mitch On Mon, Feb 15, 2016 at 9:54 AM, Brian West wrote: > I don't see why not, it may require more testing and some work.... > > On Mon, Feb 15, 2016 at 11:39 AM, Tamas Jalsovszky > wrote: > >> Are you going to support upcoming 16.04 LTS too? >> >> Jalsot >> >> On Mon, Feb 15, 2016 at 5:03 PM, Brian West wrote: >> >>> For those interested https://www.gofundme.com/freeswitch_ubuntu >>> >>> We are working on Official packages, if you care to donate to the cause. >>> >>> /b >>> >>> On Mon, Feb 15, 2016 at 9:51 AM, Mitch Capper >>> wrote: >>> >>>> I haven't had an issue with 14 or 15 with FS with the stock modules >>>> (and a few extras I use). >>>> >>>> ~mitch >>>> >>>> On Mon, Feb 15, 2016 at 6:34 AM, Caragea Silviu >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation >>>>> issues for several modules >>>>> >>>>> Silviu >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >> > > > > -- > > *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/20160216/6b3d5013/attachment.html From stefano.favaro at edistar.com Tue Feb 16 17:31:40 2016 From: stefano.favaro at edistar.com (Stefano Favaro) Date: Tue, 16 Feb 2016 15:31:40 +0100 (CET) Subject: [Freeswitch-users] Problem with mod_spy In-Reply-To: Message-ID: <361891616.7421.1455633100900.JavaMail.root@mailserver.edistar.com> Ok, I see .... so, probably the problem is that I don't have any users already defined, they are runtime registered using xml_curl on an external server . Is there any other similar application that I can use to do the work? I mean conference or others? Thanks Stefano Favaro Sviluppo Servizi ed Applicazioni _____________________ Edistar Srl a socio unico soggetta a direzione e coordinamento di YourVoice SpA Via Artigianato 1 ? I ? 31050 Vedelago (TV) Italy Phone +39 0423 7331 ? Fax +39 0423 733133 mobile: +39 3456636386 skype: stefanofavaro www.edistar.com Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potr? essere effettuata denuncia alla competente Autorit?. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario ? vietata dall?art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. ----- Messaggio originale ----- Da: "?talo Rossi" A: "FreeSWITCH Users Help" Inviato: Marted?, 16 febbraio 2016 13:47:54 Oggetto: Re: [Freeswitch-users] Problem with mod_spy I mean your bridge string, not userspy, instead of sofia/internal/4750 at ... just use bridge(user/4750) to DIAL to that user. Your userspy args looks good On Tue, Feb 16, 2016 at 5:40 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: Thanks, I've tried with but unfortunately I have the same problem. SF. https://pastebin.freeswitch.org/24556 Da: "?talo Rossi" < italo at freeswitch.org > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Inviato: Luned?, 15 febbraio 2016 20:57:29 Oggetto: Re: [Freeswitch-users] Problem with mod_spy It's probably due to the way you're dialing to your user, I can see in your log this bridge command: bridge(sofia/internal/4750%myserver) Just try to dial bridge user/4750 it'll work if 4750 is registered with FS. On Mon, Feb 15, 2016 at 6:12 AM, Stefano Favaro < stefano.favaro at edistar.com > wrote: Thank you Italo. This is my log https://pastebin.freeswitch.org/24555 and this is the id of my user spy call 94bae5a0-dd2c-4d0e-99fd-8ef81e753227 Stefano Da: "?talo Rossi" < italo at freeswitch.org > A: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Inviato: Gioved?, 11 febbraio 2016 18:52:46 Oggetto: Re: [Freeswitch-users] Problem with mod_spy 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 < stefano.favaro at edistar.com > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160216/b418bbf5/attachment-0001.html From veerabhadrarao.kankatala at panamaxil.com Tue Feb 16 17:52:57 2016 From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala) Date: Tue, 16 Feb 2016 09:52:57 -0500 (EST) Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference. Message-ID: <914027041.7946065.1455634377236.JavaMail.zimbra@panamaxil.com> Hello, I am using freeswitch-1.6.5 version for webRTC I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call both are registered in freeswitch WebRTC Client: 1001 Default user GrandStream device: 1009 Default user When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing following is the log Please tell me how to resolve this issue and let me know still anything i am missing. -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala at panamaxil.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/dd42faab/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: webRtc_Dump_log Type: application/octet-stream Size: 17804 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/dd42faab/attachment-0001.obj From veerabhadrarao.kankatala at panamaxil.com Tue Feb 16 17:54:43 2016 From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala) Date: Tue, 16 Feb 2016 09:54:43 -0500 (EST) Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference. Message-ID: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com> Hello, I am using freeswitch-1.6.5 version for webRTC I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call both are registered in freeswitch WebRTC Client: 1001 Default user GrandStream device: 1009 Default user When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing Please find the attachment for Call Log to results the coreDump Please tell me how to resolve this issue and let me know still anything i am missing. -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala at panamaxil.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/7c24dca1/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: webRtc_Dump_log Type: application/octet-stream Size: 17804 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/7c24dca1/attachment-0001.obj From krice at freeswitch.org Tue Feb 16 18:00:40 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 16 Feb 2016 09:00:40 -0600 Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference. In-Reply-To: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com> References: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com> Message-ID: All crashes are bugs and all bugs go to jira https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304 Sent from my iPhone > On Feb 16, 2016, at 8:54 AM, Veerabhadrarao Kankatala wrote: > > Hello, > > I am using freeswitch-1.6.5 version for webRTC > > I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call > > both are registered in freeswitch > > WebRTC Client: 1001 Default user > GrandStream device: 1009 Default user > > When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing > > Please find the attachment for Call Log to results the coreDump > > Please tell me how to resolve this issue and let me know still anything i am missing. > > > -- > Thanks & Regards > Veerabhadrarao Kankatala > Software Developer (C & Unix) > PANAMAX INFOTECH LIMITED > Mobile: +91-8401231249 > Messenger Id: Skype: veerabhadrarao.kankatala > E-mail: veerabhadrarao.kankatala at panamaxil.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/20160216/2afdb901/attachment.html From mike at jerris.com Tue Feb 16 18:24:24 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 10:24:24 -0500 Subject: [Freeswitch-users] 400 Bad To Header. In-Reply-To: References: Message-ID: I thought the # had to be encoded. Doesn't make sense that it wouldn't give you the error every time On Tuesday, February 16, 2016, Aqs Younas wrote: > Hi users, > > My freeswitch is replying with me 400 Bad To Header whenever It receives > Bye containing this To Header. > > To: 1111#15122031234 >;tag=NKU2jrZvFaK0j > > But I wonder in the case of Invite containing, this header everything is > fine. > > Is there something wrong with this header? (#?) > > Best regards. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/78c58159/attachment.html From mike at jerris.com Tue Feb 16 18:26:41 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 10:26:41 -0500 Subject: [Freeswitch-users] WebRTC Core Dump for Video Conference. In-Reply-To: References: <1441201861.7946338.1455634483249.JavaMail.zimbra@panamaxil.com> Message-ID: But not until you try the latest release and master to confirm we haven't already fixed it On Tuesday, February 16, 2016, Ken Rice wrote: > All crashes are bugs and all bugs go to jira > > > https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/5046304 > > Sent from my iPhone > > On Feb 16, 2016, at 8:54 AM, Veerabhadrarao Kankatala < > veerabhadrarao.kankatala at panamaxil.com > > > wrote: > > Hello, > > I am using freeswitch-1.6.5 version for webRTC > > I am using Chrome to intiate call (WebRTC Client) to another Video device > (Grand stream), and i am using Default Dialplan to make call > > both are registered in freeswitch > > WebRTC Client: 1001 Default user > GrandStream device: 1009 Default user > > When i make audio calling it is working fine, but when i make Video call > from WebRTC Client ----> GrandStream device, my freeswitch is crashing > > Please find the attachment for Call Log to results the coreDump > > Please tell me how to resolve this issue and let me know still anything i > am missing. > > > -- > Thanks & Regards > *Veerabhadrarao Kankatala* > Software Developer (C & Unix) > *PANAMAX INFOTECH LIMITED* > *Mobile:* *+91-8401231249* > *Messenger Id:* Skype: veerabhadrarao.kankatala > *E-mail: **veerabhadrarao.kankatala at panamaxil.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/20160216/fb8bf34c/attachment.html From tarantul at gmail.com Tue Feb 16 14:52:44 2016 From: tarantul at gmail.com (Nick 'tarantul' Novikov) Date: Tue, 16 Feb 2016 14:52:44 +0300 Subject: [Freeswitch-users] no audio after hold Message-ID: Hello I have some problem with sound after hold usage. Sometime the sound disappear after callcenter agent shift unhold.For callcenter agents we use sip.js (version 0.7.2). In freeswitch logs I see strange lines ( https://pastebin.freeswitch.org/24557). Can anyone explain, what happened with audio and how to fix it? -- tarantul Dios es Amor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/1db434eb/attachment-0001.html From roman.kudinov at novelapp.com Tue Feb 16 15:40:58 2016 From: roman.kudinov at novelapp.com (Roman Kudinov) Date: Tue, 16 Feb 2016 15:40:58 +0300 Subject: [Freeswitch-users] problems with bridging a call, looks like transcoding is disabled Message-ID: <56C318DA.7060603@novelapp.com> Hi all, I have a problem with bridging a call. My FS 1.6.6 is setup to bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway. I have two branches in the dial plan. 1) One works through mod_conference which calls an outbound number using conference_set_auto_outcall 2) Another works by the direct bridging of incoming rtmp call into outbound SIP call. Whilst the first branch works just fine, the second one does not. They both use the same sofia profiles, SIP gateways and outbound SIP numbers. They both are called from the same RTMP source. Here are the snippet of codes. ================== This one works ==================================== > > > > > > > > > > =================================================== ================ This one does not work ================ > > > > > > > ======================== I'd like to outline that they use the same SIP profiles, they are called from the same RTMP-source (they differs by the destination_number), they call the same SIP number. I turned on SIP tracing on and found that the call that is initiated by mod_conference offers the codecs according to outbound_codec_prefs set in vars.xml, here is the piece of log: > m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110 > a=rtpmap:0 PCMU/8000 > a=rtpmap:102 SPEEX/8000 > a=rtpmap:103 SPEEX/16000 > a=rtpmap:104 SPEEX/32000 > a=rtpmap:105 opus/48000/2 > a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:106 telephone-event/16000 > a=fmtp:106 0-16 > a=rtpmap:108 telephone-event/32000 > a=fmtp:108 0-16 > a=rtpmap:110 telephone-event/48000 > a=fmtp:110 0-16 > a=ptime:20 But the directly bridged call offers incoming codec only, e.g. speex > m=audio 24972 RTP/AVP 102 101 > a=rtpmap:102 SPEEX/16000 > a=rtpmap:101 telephone-event/16000 > a=fmtp:101 0-16 > a=ptime:20 I tried everything I could imagine. I set absolute_codec_string in the dialplan (you can see it in the above snippet). I explicitly set > in vars.xml I tried to change > from generous to greedy I tried with true/false in the following parameters in internal.xml and external.xml SOFIA profiles > > Nothing changes. Moreover the direct bridging worked fine on FS 1.4.7, it offered PCMU and Speex codecs to SIP. I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I missed an important setting that makes "bridge" application to work in proxy media mode. I checked I don't have neither bypass or proxy words in vars.xml, sofia.conf.xml, internal.xml, external.xml, public.xml or they are commented. Does anybody have any ideas about the reason for such behavior? Thanks, Roman From aqsyounas at gmail.com Tue Feb 16 18:39:25 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 16 Feb 2016 20:39:25 +0500 Subject: [Freeswitch-users] 400 Bad To Header. In-Reply-To: References: Message-ID: Thanks Michael, What how can i encode this pound key? any hint is much appreciated. Best Regards This email has been sent from a virus-free computer protected by Avast. www.avast.com <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On 16 February 2016 at 20:24, Michael Jerris wrote: > I thought the # had to be encoded. Doesn't make sense that it wouldn't > give you the error every time > > > On Tuesday, February 16, 2016, Aqs Younas wrote: > >> Hi users, >> >> My freeswitch is replying with me 400 Bad To Header whenever It receives >> Bye containing this To Header. >> >> To: 1111#15122031234 > >;tag=NKU2jrZvFaK0j >> >> But I wonder in the case of Invite containing, this header everything is >> fine. >> >> Is there something wrong with this header? (#?) >> >> 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/20160216/4e04b948/attachment.html From brian at freeswitch.org Tue Feb 16 19:00:46 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Feb 2016 10:00:46 -0600 Subject: [Freeswitch-users] 400 Bad To Header. In-Reply-To: References: Message-ID: Its because its not quoted. To: "1111#15122031234" ;tag=NKU2jrZvFaK0j would be correct, have your upstream FIX their broken stuff. /b On Tue, Feb 16, 2016 at 9:39 AM, Aqs Younas wrote: > Thanks Michael, > > What how can i encode this pound key? any hint is much appreciated. > > Best Regards > > This email has been sent from a virus-free computer protected by Avast. > www.avast.com > <#-734131114_DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > On 16 February 2016 at 20:24, Michael Jerris wrote: > >> I thought the # had to be encoded. Doesn't make sense that it wouldn't >> give you the error every time >> >> >> On Tuesday, February 16, 2016, Aqs Younas wrote: >> >>> Hi users, >>> >>> My freeswitch is replying with me 400 Bad To Header whenever It receives >>> Bye containing this To Header. >>> >>> To: 1111#15122031234 >> >;tag=NKU2jrZvFaK0j >>> >>> But I wonder in the case of Invite containing, this header everything is >>> fine. >>> >>> Is there something wrong with this header? (#?) >>> >>> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/e97c70f5/attachment.html From mike at jerris.com Tue Feb 16 19:02:02 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 11:02:02 -0500 Subject: [Freeswitch-users] 400 Bad To Header. In-Reply-To: References: Message-ID: You'll have to fix that on your device... > On Feb 16, 2016, at 10:39 AM, Aqs Younas wrote: > > Thanks Michael, > > What how can i encode this pound key? any hint is much appreciated. > > Best Regards > > This email has been sent from a virus-free computer protected by Avast. > www.avast.com > > On 16 February 2016 at 20:24, Michael Jerris > wrote: > I thought the # had to be encoded. Doesn't make sense that it wouldn't give you the error every time > > > On Tuesday, February 16, 2016, Aqs Younas > wrote: > Hi users, > > My freeswitch is replying with me 400 Bad To Header whenever It receives Bye containing this To Header. > > To: 1111#15122031234 >;tag=NKU2jrZvFaK0j > > But I wonder in the case of Invite containing, this header everything is fine. > > Is there something wrong with this header? (#?) > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/b3bec7ba/attachment-0001.html From gmaruzz at gmail.com Tue Feb 16 19:06:57 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 16 Feb 2016 17:06:57 +0100 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: Please pastebin a complete sip trace (from fs-cli: sofia global siptrace on) and then put here the link to pastebin. Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" ha scritto: > Hello > > I have some problem with sound after hold usage. Sometime the sound > disappear after callcenter agent shift unhold.For callcenter agents we use > sip.js (version 0.7.2). > In freeswitch logs I see strange lines ( > https://pastebin.freeswitch.org/24557). > Can anyone explain, what happened with audio and how to fix it? > > > -- > tarantul > Dios es Amor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/b0858ad6/attachment.html From aqsyounas at gmail.com Tue Feb 16 19:07:26 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 16 Feb 2016 21:07:26 +0500 Subject: [Freeswitch-users] 400 Bad To Header. In-Reply-To: References: Message-ID: Much Thanks :) On 16-Feb-2016 9:04 pm, "Michael Jerris" wrote: > You'll have to fix that on your device... > > On Feb 16, 2016, at 10:39 AM, Aqs Younas wrote: > > Thanks Michael, > > What how can i encode this pound key? any hint is much appreciated. > > Best Regards > > This email has been sent from a virus-free computer protected by Avast. > www.avast.com > > On 16 February 2016 at 20:24, Michael Jerris wrote: > >> I thought the # had to be encoded. Doesn't make sense that it wouldn't >> give you the error every time >> >> >> On Tuesday, February 16, 2016, Aqs Younas wrote: >> >>> Hi users, >>> >>> My freeswitch is replying with me 400 Bad To Header whenever It receives >>> Bye containing this To Header. >>> >>> To: 1111#15122031234 >> >;tag=NKU2jrZvFaK0j >>> >>> But I wonder in the case of Invite containing, this header everything is >>> fine. >>> >>> Is there something wrong with this header? (#?) >>> >>> 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/3522aa5f/attachment.html From brian at freeswitch.org Tue Feb 16 19:11:47 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Feb 2016 10:11:47 -0600 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: Your logs are incomplete, need full debug logs. Please don't filter them excessively and try on master please. On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli wrote: > Please pastebin a complete sip trace (from fs-cli: sofia global siptrace > on) and then put here the link to pastebin. > Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" ha > scritto: > >> Hello >> >> I have some problem with sound after hold usage. Sometime the sound >> disappear after callcenter agent shift unhold.For callcenter agents we use >> sip.js (version 0.7.2). >> In freeswitch logs I see strange lines ( >> https://pastebin.freeswitch.org/24557). >> Can anyone explain, what happened with audio and how to fix it? >> >> >> -- >> tarantul >> Dios es Amor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160216/5b209c5f/attachment.html From brian at freeswitch.org Tue Feb 16 19:15:42 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Feb 2016 10:15:42 -0600 Subject: [Freeswitch-users] problems with bridging a call, looks like transcoding is disabled In-Reply-To: <56C318DA.7060603@novelapp.com> References: <56C318DA.7060603@novelapp.com> Message-ID: This topic was talked about on the list in the past week: https://freeswitch.org/jira/browse/FS-8321 "BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix inbound and outbound codecs" /b On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov wrote: > Hi all, > > I have a problem with bridging a call. My FS 1.6.6 is setup to bridge > calls from RTMP-based source (using mod_rtmp) to a SIP gateway. > I have two branches in the dial plan. > > 1) One works through mod_conference which calls an outbound number using > conference_set_auto_outcall > > 2) Another works by the direct bridging of incoming rtmp call into > outbound SIP call. > > Whilst the first branch works just fine, the second one does not. They > both use the same sofia profiles, SIP gateways and outbound SIP numbers. > They both are called from the same RTMP source. Here are the snippet of > codes. > > ================== This one works ==================================== > > > > > > > > data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> > > data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> > > data="conference_auto_outcall_profile=default"/> > > data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/> > > data="$1+flags{moderator|endconf|mute}"/> > > > > > =================================================== > > ================ This one does not work ================ > > > > > > > > data="absolute_codec_string=PCMU,PCMA,opus"/> > > data="sofia/gateway/sip_profile/number"/> > > > > > ======================== > > I'd like to outline that they use the same SIP profiles, they are called > from the same RTMP-source (they differs by the destination_number), they > call the same SIP number. > I turned on SIP tracing on and found that the call that is initiated by > mod_conference offers the codecs according to outbound_codec_prefs set > in vars.xml, here is the piece of log: > > m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:102 SPEEX/8000 > > a=rtpmap:103 SPEEX/16000 > > a=rtpmap:104 SPEEX/32000 > > a=rtpmap:105 opus/48000/2 > > a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; > maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:106 telephone-event/16000 > > a=fmtp:106 0-16 > > a=rtpmap:108 telephone-event/32000 > > a=fmtp:108 0-16 > > a=rtpmap:110 telephone-event/48000 > > a=fmtp:110 0-16 > > a=ptime:20 > > But the directly bridged call offers incoming codec only, e.g. speex > > m=audio 24972 RTP/AVP 102 101 > > a=rtpmap:102 SPEEX/16000 > > a=rtpmap:101 telephone-event/16000 > > a=fmtp:101 0-16 > > a=ptime:20 > > I tried everything I could imagine. I set absolute_codec_string in the > dialplan (you can see it in the above snippet). > I explicitly set > > data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/> > in vars.xml > > I tried to change > > > from generous to greedy > > I tried with true/false in the following parameters in internal.xml and > external.xml SOFIA profiles > > > > > Nothing changes. > > Moreover the direct bridging worked fine on FS 1.4.7, it offered PCMU and > Speex codecs to SIP. > I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I missed > an important setting that makes "bridge" application to work in > proxy media mode. > I checked I don't have neither bypass or proxy words in vars.xml, > sofia.conf.xml, internal.xml, external.xml, public.xml or they are > commented. > > Does anybody have any ideas about the reason for such behavior? > > > Thanks, > 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 > -- *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/20160216/4a12706a/attachment-0001.html From decipher.hk at gmail.com Tue Feb 16 19:40:29 2016 From: decipher.hk at gmail.com (=?utf-8?B?Um9kcmlnbyBSYW3DrXJleiBOb3JhbWJ1ZW5h?=) Date: Tue, 16 Feb 2016 16:40:29 +0000 Subject: [Freeswitch-users] Answered/abandoned calls mod_callcenter In-Reply-To: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com> References: <27e84038904cd4a54ac63be19844adb2@mail2.boxtub.com> <701581496dd4988b32ee3289cc2166df@mail2.boxtub.com> Message-ID: February 7 2016 3:18 PM, "Rodrigo Ram?rez Norambuena" wrote: > 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 Hi everyone, I added support for QPanel of FreeSWITCH :) You can test the branch fs into https://github.com/roramirez/qpanel/tree/fs If you have any issue or improvement is welcome to hear. Regards, -- Rodrigo Ram?rez Norambuena http://www.rodrigoramirez.com From nneul at mst.edu Tue Feb 16 20:20:06 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 16 Feb 2016 11:20:06 -0600 Subject: [Freeswitch-users] verto jsonrpc settings? Message-ID: <56C35A46.2080509@mst.edu> What does the verto/directory setting: ' expose? It's very unclear from documentation. Does this make ALL conference/presence events available to that user that logged in via verto or only those related to their particular registration/call? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From roman.kudinov at novelapp.com Tue Feb 16 20:53:41 2016 From: roman.kudinov at novelapp.com (Roman Kudinov) Date: Tue, 16 Feb 2016 20:53:41 +0300 Subject: [Freeswitch-users] problems with bridging a call, looks like transcoding is disabled In-Reply-To: References: Message-ID: <56C36225.4080906@novelapp.com> Hi Brian, I saw this variable but looks like used it the wrong way. Thanks for the help, the codecs negotiation works properly now! 16.02.2016 19:16, Brian West: > This topic was talked about on the list in the past week: > https://freeswitch.org/jira/browse/FS-8321 > > "BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix > inbound and outbound codecs" > > /b > > On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov > > wrote: > > Hi all, > > I have a problem with bridging a call. My FS 1.6.6 is setup to > bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway. > I have two branches in the dial plan. > > 1) One works through mod_conference which calls an outbound number > using conference_set_auto_outcall > > 2) Another works by the direct bridging of incoming rtmp call into > outbound SIP call. > > Whilst the first branch works just fine, the second one does not. > They both use the same sofia profiles, SIP gateways and outbound > SIP numbers. > They both are called from the same RTMP source. Here are the > snippet of codes. > > ================== This one works ==================================== > > > > > > > > data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> > > data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> > > data="conference_auto_outcall_profile=default"/> > > data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/> > > data="$1+flags{moderator|endconf|mute}"/> > > > > > =================================================== > > ================ This one does not work ================ > > > > > > > > data="absolute_codec_string=PCMU,PCMA,opus"/> > > data="sofia/gateway/sip_profile/number"/> > > > > > ======================== > > I'd like to outline that they use the same SIP profiles, they are > called from the same RTMP-source (they differs by the > destination_number), they > call the same SIP number. > I turned on SIP tracing on and found that the call that is > initiated by mod_conference offers the codecs according to > outbound_codec_prefs set > in vars.xml, here is the piece of log: > > m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:102 SPEEX/8000 > > a=rtpmap:103 SPEEX/16000 > > a=rtpmap:104 SPEEX/32000 > > a=rtpmap:105 opus/48000/2 > > a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; > maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:106 telephone-event/16000 > > a=fmtp:106 0-16 > > a=rtpmap:108 telephone-event/32000 > > a=fmtp:108 0-16 > > a=rtpmap:110 telephone-event/48000 > > a=fmtp:110 0-16 > > a=ptime:20 > > But the directly bridged call offers incoming codec only, e.g. speex > > m=audio 24972 RTP/AVP 102 101 > > a=rtpmap:102 SPEEX/16000 > > a=rtpmap:101 telephone-event/16000 > > a=fmtp:101 0-16 > > a=ptime:20 > > I tried everything I could imagine. I set absolute_codec_string in > the dialplan (you can see it in the above snippet). > I explicitly set > > data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/> > in vars.xml > > I tried to change > > > from generous to greedy > > I tried with true/false in the following parameters in > internal.xml and > external.xml SOFIA profiles > > > > > Nothing changes. > > Moreover the direct bridging worked fine on FS 1.4.7, it offered > PCMU and Speex codecs to SIP. > I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I > missed an important setting that makes "bridge" application to work in > proxy media mode. > I checked I don't have neither bypass or proxy words in vars.xml, > sofia.conf.xml, internal.xml, external.xml, public.xml or they are > commented. > > Does anybody have any ideas about the reason for such behavior? > > > Thanks, > 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 > > > > > -- > > */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/20160216/fc09b70c/attachment.html From blackc2004 at gmail.com Tue Feb 16 22:14:58 2016 From: blackc2004 at gmail.com (Cj B) Date: Tue, 16 Feb 2016 11:14:58 -0800 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal Message-ID: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> Hi everyone, I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal http://pastebin.com/nxi9bH0N Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it? uname -a Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux /usr/local/freeswitch/bin/freeswitch -version FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) Thanks, Cj B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/2d13670b/attachment-0001.html From abaci64 at gmail.com Tue Feb 16 22:15:27 2016 From: abaci64 at gmail.com (Abaci B) Date: Tue, 16 Feb 2016 14:15:27 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: I remember in the early days of FreeSWITCH it was recommended to turn off hyperthreading as it had a negative impact on FreeSWITCH performace, Can someone confirm if this has changed with newer processors or newer kernel, if hyperthreading boosts performance then I would guess that a 10 core xeon would give me at least as much performace as a 16 core AMD. Thanks On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris wrote: > You would have to test to see, most of my experience is a couple chip > generations back now but bang for the buck was much better with intel then > and we saw many more weird issues on AMD boxes. > > > On Sunday, February 14, 2016, Abaci B wrote: > >> Hi all, >> I know that in general Intel Xeon processors are recommended over AMD for >> FreeSWITCH, my question is if AMD is really bad or just not as good as >> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about >> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I >> go for the AMD which will give me 4x the amount of cores or stick to Xeon, >> my goal is to get the maximum amount of channels/calls (doing lua IVR with >> no transcoding) and conferencing. >> Any help, feedback, benchmarks or personal experience would be appreciated >> 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/20160216/5a4757ca/attachment.html From gmaruzz at gmail.com Tue Feb 16 22:25:52 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 16 Feb 2016 20:25:52 +0100 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal In-Reply-To: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> Message-ID: Why you want to stop a debug message? Il 16/Feb/2016 20:16, "Cj B" ha scritto: > Hi everyone, > > I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the > following repeat over and over in the logs. > > 1. > > 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal > 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal > > http://pastebin.com/nxi9bH0N > > Calls appear to be completing properly and phones are registered. I was > just wondering what this means and if there?s a way to stop it? > > uname -a > Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian > 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux > > /usr/local/freeswitch/bin/freeswitch -version > FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git > d2d0b32 2016-01-11 20:16:12Z 64bit) > > Thanks, > Cj B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/e364adab/attachment.html From mike at jerris.com Tue Feb 16 22:48:08 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 14:48:08 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: Have not tested it lately, but likely still true, in regards to "I guess"... why not just try it and see for sure? > On Feb 16, 2016, at 2:15 PM, Abaci B wrote: > > I remember in the early days of FreeSWITCH it was recommended to turn off hyperthreading as it had a negative impact on FreeSWITCH performace, Can someone confirm if this has changed with newer processors or newer kernel, if hyperthreading boosts performance then I would guess that a 10 core xeon would give me at least as much performace as a 16 core AMD. > Thanks > > On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris > wrote: > You would have to test to see, most of my experience is a couple chip generations back now but bang for the buck was much better with intel then and we saw many more weird issues on AMD boxes. > > > On Sunday, February 14, 2016, Abaci B > wrote: > Hi all, > I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls (doing lua IVR with no transcoding) and conferencing. > Any help, feedback, benchmarks or personal experience would be appreciated > Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/8f262335/attachment.html From blackc2004 at gmail.com Tue Feb 16 22:50:02 2016 From: blackc2004 at gmail.com (Cj B) Date: Tue, 16 Feb 2016 11:50:02 -0800 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal In-Reply-To: References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> Message-ID: I would like to know what it means so that if it?s a problem I can fix it. Thanks Cj B > On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli wrote: > > Why you want to stop a debug message? > > Il 16/Feb/2016 20:16, "Cj B" > ha scritto: > Hi everyone, > > I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. > 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal > 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal > http://pastebin.com/nxi9bH0N > Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it? > > uname -a > Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux > > /usr/local/freeswitch/bin/freeswitch -version > FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) > > Thanks, > Cj B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/ce55e5b1/attachment-0001.html From mike at jerris.com Tue Feb 16 22:55:23 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 14:55:23 -0500 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal In-Reply-To: References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> Message-ID: <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com> From the block of code: if (is_dialog) { // Usually we report the dialogs FROM the probed user. The exception is when the monitored endpoint is internal, // and its presence_id is set in the dialplan. Reverse the direction if this is not a registered entity. const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username")); if (!strcmp(direction, "inbound") && strcmp(sub_to_user, caller)) { // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed // because it is not going through the B2BUA switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n"); direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound"; } } This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before. Are you actually having something not working? > On Feb 16, 2016, at 2:50 PM, Cj B wrote: > > I would like to know what it means so that if it?s a problem I can fix it. > > Thanks > Cj B > >> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli > wrote: >> >> Why you want to stop a debug message? >> >> Il 16/Feb/2016 20:16, "Cj B" > ha scritto: >> Hi everyone, >> >> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. >> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >> http://pastebin.com/nxi9bH0N >> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it? >> >> uname -a >> Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux >> >> /usr/local/freeswitch/bin/freeswitch -version >> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) >> >> Thanks, >> Cj B >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160216/8ca7be8d/attachment.html From abaci64 at gmail.com Tue Feb 16 23:01:35 2016 From: abaci64 at gmail.com (Abaci B) Date: Tue, 16 Feb 2016 15:01:35 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L (40 cores/80 threads), I wish I would be able to afford both and see which one (AMD vs Intel) is better... so if I know that hyperthreading is boosting performance I would probably stick to intel as in general it's better and it's more cores. On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris wrote: > Have not tested it lately, but likely still true, in regards to "I > guess"... why not just try it and see for sure? > > On Feb 16, 2016, at 2:15 PM, Abaci B wrote: > > I remember in the early days of FreeSWITCH it was recommended to turn off > hyperthreading as it had a negative impact on FreeSWITCH performace, Can > someone confirm if this has changed with newer processors or newer kernel, > if hyperthreading boosts performance then I would guess that a 10 core xeon > would give me at least as much performace as a 16 core AMD. > Thanks > > On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris wrote: > >> You would have to test to see, most of my experience is a couple chip >> generations back now but bang for the buck was much better with intel then >> and we saw many more weird issues on AMD boxes. >> >> >> On Sunday, February 14, 2016, Abaci B wrote: >> >>> Hi all, >>> I know that in general Intel Xeon processors are recommended over AMD >>> for FreeSWITCH, my question is if AMD is really bad or just not as good as >>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about >>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I >>> go for the AMD which will give me 4x the amount of cores or stick to Xeon, >>> my goal is to get the maximum amount of channels/calls (doing lua IVR with >>> no transcoding) and conferencing. >>> Any help, feedback, benchmarks or personal experience would be >>> appreciated >>> 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/20160216/062e2cd1/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 16 23:07:36 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Feb 2016 14:07:36 -0600 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: If you want an opinionated answer to help you decide: Always use Intel. AMD is Intel's R&D department. The performance is better and the hyperthreading on modern Intel is fine. On Tue, Feb 16, 2016 at 2:01 PM, Abaci B wrote: > I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L > (40 cores/80 threads), I wish I would be able to afford both and see which > one (AMD vs Intel) is better... so if I know that hyperthreading is > boosting performance I would probably stick to intel as in general it's > better and it's more cores. > > On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris wrote: > >> Have not tested it lately, but likely still true, in regards to "I >> guess"... why not just try it and see for sure? >> >> On Feb 16, 2016, at 2:15 PM, Abaci B wrote: >> >> I remember in the early days of FreeSWITCH it was recommended to turn off >> hyperthreading as it had a negative impact on FreeSWITCH performace, Can >> someone confirm if this has changed with newer processors or newer kernel, >> if hyperthreading boosts performance then I would guess that a 10 core xeon >> would give me at least as much performace as a 16 core AMD. >> Thanks >> >> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris wrote: >> >>> You would have to test to see, most of my experience is a couple chip >>> generations back now but bang for the buck was much better with intel then >>> and we saw many more weird issues on AMD boxes. >>> >>> >>> On Sunday, February 14, 2016, Abaci B wrote: >>> >>>> Hi all, >>>> I know that in general Intel Xeon processors are recommended over AMD >>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as >>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about >>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I >>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon, >>>> my goal is to get the maximum amount of channels/calls (doing lua IVR with >>>> no transcoding) and conferencing. >>>> Any help, feedback, benchmarks or personal experience would be >>>> appreciated >>>> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/8060a237/attachment.html From blackc2004 at gmail.com Tue Feb 16 23:07:32 2016 From: blackc2004 at gmail.com (Cj B) Date: Tue, 16 Feb 2016 12:07:32 -0800 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal In-Reply-To: <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com> References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com> Message-ID: <3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com> Thanks Michael. So far I don?t see anything not working but it?s only been a few hours since I turned on this server to start accepting real traffic. But based on this code, it looks to me like it should only be happening during a call, correct? It looks to me like I?m getting this log repeating even when there is no call traffic on the server, like it?s happening on phone register. Thanks Cj B > On Feb 16, 2016, at 11:55 AM, Michael Jerris wrote: > > From the block of code: > > if (is_dialog) { > // Usually we report the dialogs FROM the probed user. The exception is when the monitored endpoint is internal, > // and its presence_id is set in the dialplan. Reverse the direction if this is not a registered entity. > const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username")); > if (!strcmp(direction, "inbound") && strcmp(sub_to_user, caller)) { > // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed > // because it is not going through the B2BUA > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n"); > direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound"; > } > > } > > This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before. Are you actually having something not working? > >> On Feb 16, 2016, at 2:50 PM, Cj B > wrote: >> >> I would like to know what it means so that if it?s a problem I can fix it. >> >> Thanks >> Cj B >> >>> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli > wrote: >>> >>> Why you want to stop a debug message? >>> >>> Il 16/Feb/2016 20:16, "Cj B" > ha scritto: >>> Hi everyone, >>> >>> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. >>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >>> http://pastebin.com/nxi9bH0N >>> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it? >>> >>> uname -a >>> Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux >>> >>> /usr/local/freeswitch/bin/freeswitch -version >>> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) >>> >>> Thanks, >>> Cj B >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160216/bf73d554/attachment-0001.html From abaci64 at gmail.com Tue Feb 16 23:13:51 2016 From: abaci64 at gmail.com (Abaci B) Date: Tue, 16 Feb 2016 15:13:51 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be considered good for hyperthreading? On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you want an opinionated answer to help you decide: > > Always use Intel. AMD is Intel's R&D department. The performance is > better and the hyperthreading on modern Intel is fine. > > > On Tue, Feb 16, 2016 at 2:01 PM, Abaci B wrote: > >> I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L >> (40 cores/80 threads), I wish I would be able to afford both and see which >> one (AMD vs Intel) is better... so if I know that hyperthreading is >> boosting performance I would probably stick to intel as in general it's >> better and it's more cores. >> >> On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris wrote: >> >>> Have not tested it lately, but likely still true, in regards to "I >>> guess"... why not just try it and see for sure? >>> >>> On Feb 16, 2016, at 2:15 PM, Abaci B wrote: >>> >>> I remember in the early days of FreeSWITCH it was recommended to turn >>> off hyperthreading as it had a negative impact on FreeSWITCH performace, >>> Can someone confirm if this has changed with newer processors or newer >>> kernel, if hyperthreading boosts performance then I would guess that a 10 >>> core xeon would give me at least as much performace as a 16 core AMD. >>> Thanks >>> >>> On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris >>> wrote: >>> >>>> You would have to test to see, most of my experience is a couple chip >>>> generations back now but bang for the buck was much better with intel then >>>> and we saw many more weird issues on AMD boxes. >>>> >>>> >>>> On Sunday, February 14, 2016, Abaci B wrote: >>>> >>>>> Hi all, >>>>> I know that in general Intel Xeon processors are recommended over AMD >>>>> for FreeSWITCH, my question is if AMD is really bad or just not as good as >>>>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about >>>>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I >>>>> go for the AMD which will give me 4x the amount of cores or stick to Xeon, >>>>> my goal is to get the maximum amount of channels/calls (doing lua IVR with >>>>> no transcoding) and conferencing. >>>>> Any help, feedback, benchmarks or personal experience would be >>>>> appreciated >>>>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/097ccaeb/attachment.html From mike at jerris.com Tue Feb 16 23:25:50 2016 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Feb 2016 15:25:50 -0500 Subject: [Freeswitch-users] repeating: sofia_presence_sub_callback: endpt is internal In-Reply-To: <3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com> References: <54FDCCCA-1198-4513-A259-1D2ED84B68FA@gmail.com> <7906696F-9366-4697-98BB-97EC561ED1F5@jerris.com> <3706BD46-5D4D-4C3B-97E3-2821B5B7F7C3@gmail.com> Message-ID: its related to presence code... > On Feb 16, 2016, at 3:07 PM, Cj B wrote: > > Thanks Michael. So far I don?t see anything not working but it?s only been a few hours since I turned on this server to start accepting real traffic. > > But based on this code, it looks to me like it should only be happening during a call, correct? It looks to me like I?m getting this log repeating even when there is no call traffic on the server, like it?s happening on phone register. > > Thanks > Cj B > >> On Feb 16, 2016, at 11:55 AM, Michael Jerris > wrote: >> >> From the block of code: >> >> if (is_dialog) { >> // Usually we report the dialogs FROM the probed user. The exception is when the monitored endpoint is internal, >> // and its presence_id is set in the dialplan. Reverse the direction if this is not a registered entity. >> const char *caller = switch_str_nil(switch_event_get_header(helper->event, "caller-username")); >> if (!strcmp(direction, "inbound") && strcmp(sub_to_user, caller)) { >> // If inbound and the entity is not the caller (i.e. internal to FS), then the direction is reversed >> // because it is not going through the B2BUA >> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "sofia_presence_sub_callback: endpt is internal\n"); >> direction = !strcasecmp(direction, "outbound") ? "inbound" : "outbound"; >> } >> >> } >> >> This is from a patch from 2011, and would have been in 1.4.26 as well, so you probably just didn't notice it before. Are you actually having something not working? >> >>> On Feb 16, 2016, at 2:50 PM, Cj B > wrote: >>> >>> I would like to know what it means so that if it?s a problem I can fix it. >>> >>> Thanks >>> Cj B >>> >>>> On Feb 16, 2016, at 11:25 AM, Giovanni Maruzzelli > wrote: >>>> >>>> Why you want to stop a debug message? >>>> >>>> Il 16/Feb/2016 20:16, "Cj B" > ha scritto: >>>> Hi everyone, >>>> >>>> I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. >>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >>>> 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal >>>> http://pastebin.com/nxi9bH0N >>>> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there?s a way to stop it? >>>> >>>> uname -a >>>> Linux my.domainname.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux >>>> >>>> /usr/local/freeswitch/bin/freeswitch -version >>>> FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) >>>> >>>> Thanks, >>>> Cj B >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20160216/ac6bebf3/attachment-0001.html From mgg at giagnocavo.net Tue Feb 16 23:47:46 2016 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 16 Feb 2016 20:47:46 +0000 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: If you?re actually doing one huge conference so you need all calls into a single FS instance, ignore this message. I might be a broken record on this topic, but seriously consider running multiple FreeSWITCH instances and setting affinity to specific sockets or even groups of cores. I?d be surprised to hear a quad-socket system is price/perf competitive with multiple smaller boxes. At a minimum, you want to partition on NUMA nodes, otherwise every single memory access needs to jump through other processors. For instance, if CPU0 wants memory that?s connected to CPU3, it needs to request it via CPU1 then CPU3. This is slow (a single hop introduces a ~40% penalty). After that, anytime threads are writing to the same part of memory, CPUs have to take ownership from each other?s caches. This can be as expensive as going to main memory (thus being as slow as having no cache) and kill perf. Presumably with enough media load, you?d want to look into how your NIC is delivering packets, and distribute that accordingly. On Xeon 5500 series (a while ago, on FS 1.2) we didn?t notice any real impact of hyperthreading either way. But keeping FS partitioned on groups of 4 threads (2 cores) seemed to be the best performing option. The increase in stability was very marked (versus one FS process across 2x Xeons, 16 threads). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci B Sent: Tuesday, 16 February, 2016 14:14 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] AMD is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be considered good for hyperthreading? On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale > wrote: If you want an opinionated answer to help you decide: Always use Intel. AMD is Intel's R&D department. The performance is better and the hyperthreading on modern Intel is fine. On Tue, Feb 16, 2016 at 2:01 PM, Abaci B > wrote: I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L (40 cores/80 threads), I wish I would be able to afford both and see which one (AMD vs Intel) is better... so if I know that hyperthreading is boosting performance I would probably stick to intel as in general it's better and it's more cores. On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris > wrote: Have not tested it lately, but likely still true, in regards to "I guess"... why not just try it and see for sure? On Feb 16, 2016, at 2:15 PM, Abaci B > wrote: I remember in the early days of FreeSWITCH it was recommended to turn off hyperthreading as it had a negative impact on FreeSWITCH performace, Can someone confirm if this has changed with newer processors or newer kernel, if hyperthreading boosts performance then I would guess that a 10 core xeon would give me at least as much performace as a 16 core AMD. Thanks On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris > wrote: You would have to test to see, most of my experience is a couple chip generations back now but bang for the buck was much better with intel then and we saw many more weird issues on AMD boxes. On Sunday, February 14, 2016, Abaci B > wrote: Hi all, I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls (doing lua IVR with no transcoding) and conferencing. Any help, feedback, benchmarks or personal experience would be appreciated 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160216/039cc7b6/attachment-0001.html From brian at freeswitch.org Wed Feb 17 00:00:02 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Feb 2016 15:00:02 -0600 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: Early hyper-threading would have been an issue, but newer CPUs seem to do a great job of it. On Tue, Feb 16, 2016 at 1:48 PM, Michael Jerris wrote: > Have not tested it lately, but likely still true, in regards to "I > guess"... why not just try it and see for sure? > > On Feb 16, 2016, at 2:15 PM, Abaci B wrote: > > I remember in the early days of FreeSWITCH it was recommended to turn off > hyperthreading as it had a negative impact on FreeSWITCH performace, Can > someone confirm if this has changed with newer processors or newer kernel, > if hyperthreading boosts performance then I would guess that a 10 core xeon > would give me at least as much performace as a 16 core AMD. > Thanks > > On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris wrote: > >> You would have to test to see, most of my experience is a couple chip >> generations back now but bang for the buck was much better with intel then >> and we saw many more weird issues on AMD boxes. >> >> >> On Sunday, February 14, 2016, Abaci B wrote: >> >>> Hi all, >>> I know that in general Intel Xeon processors are recommended over AMD >>> for FreeSWITCH, my question is if AMD is really bad or just not as good as >>> Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about >>> the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I >>> go for the AMD which will give me 4x the amount of cores or stick to Xeon, >>> my goal is to get the maximum amount of channels/calls (doing lua IVR with >>> no transcoding) and conferencing. >>> Any help, feedback, benchmarks or personal experience would be >>> appreciated >>> 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 > -- *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/20160216/a835a0bd/attachment.html From abaci64 at gmail.com Wed Feb 17 00:03:59 2016 From: abaci64 at gmail.com (Abaci B) Date: Tue, 16 Feb 2016 16:03:59 -0500 Subject: [Freeswitch-users] AMD In-Reply-To: References: Message-ID: Running one huge conference is actually part of the reason that I want to go with a huge box, but your input is actually very informative as I do intend to use multiple instance of freeswitch (using proxmox / lxc) yet still have the ability on occasion to allocate lots of CPU to a single instance. You're right that multiple smaller machines will probably win in price/performance, colo space is also a factor. Thanks Anthony and MIchael for all the info. once I have server up and running I will try to test the limits and report back. On Tue, Feb 16, 2016 at 3:47 PM, Michael Giagnocavo wrote: > If you?re actually doing one huge conference so you need all calls into a > single FS instance, ignore this message. > > > > I might be a broken record on this topic, but seriously consider running > multiple FreeSWITCH instances and setting affinity to specific sockets or > even groups of cores. I?d be surprised to hear a quad-socket system is > price/perf competitive with multiple smaller boxes. > > > > At a minimum, you want to partition on NUMA nodes, otherwise every single > memory access needs to jump through other processors. For instance, if CPU0 > wants memory that?s connected to CPU3, it needs to request it via CPU1 then > CPU3. This is slow (a single hop introduces a ~40% penalty). After that, > anytime threads are writing to the same part of memory, CPUs have to take > ownership from each other?s caches. This can be as expensive as going to > main memory (thus being as slow as having no cache) and kill perf. > > > > Presumably with enough media load, you?d want to look into how your NIC is > delivering packets, and distribute that accordingly. > > > > On Xeon 5500 series (a while ago, on FS 1.2) we didn?t notice any real > impact of hyperthreading either way. But keeping FS partitioned on groups > of 4 threads (2 cores) seemed to be the best performing option. The > increase in stability was very marked (versus one FS process across 2x > Xeons, 16 threads). > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B > *Sent:* Tuesday, 16 February, 2016 14:14 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] AMD > > > > is the E7-8867L (Westmere / LGA1567 / Apr/3/11) modern/new enough to be > considered good for hyperthreading? > > > > On Tue, Feb 16, 2016 at 3:07 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > If you want an opinionated answer to help you decide: > > > > Always use Intel. AMD is Intel's R&D department. The performance is > better and the hyperthreading on modern Intel is fine. > > > > > > On Tue, Feb 16, 2016 at 2:01 PM, Abaci B wrote: > > I'm debating between a 4x AMD 6276 (64 cores total) and 4x Intel E7-8867L > (40 cores/80 threads), I wish I would be able to afford both and see which > one (AMD vs Intel) is better... so if I know that hyperthreading is > boosting performance I would probably stick to intel as in general it's > better and it's more cores. > > > > On Tue, Feb 16, 2016 at 2:48 PM, Michael Jerris wrote: > > Have not tested it lately, but likely still true, in regards to "I > guess"... why not just try it and see for sure? > > > > On Feb 16, 2016, at 2:15 PM, Abaci B wrote: > > > > I remember in the early days of FreeSWITCH it was recommended to turn off > hyperthreading as it had a negative impact on FreeSWITCH performace, Can > someone confirm if this has changed with newer processors or newer kernel, > if hyperthreading boosts performance then I would guess that a 10 core xeon > would give me at least as much performace as a 16 core AMD. > > Thanks > > > > On Sun, Feb 14, 2016 at 12:52 PM, Michael Jerris wrote: > > You would have to test to see, most of my experience is a couple chip > generations back now but bang for the buck was much better with intel then > and we saw many more weird issues on AMD boxes. > > > > On Sunday, February 14, 2016, Abaci B wrote: > > Hi all, > > I know that in general Intel Xeon processors are recommended over AMD for > FreeSWITCH, my question is if AMD is really bad or just not as good as > Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about > the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I > go for the AMD which will give me 4x the amount of cores or stick to Xeon, > my goal is to get the maximum amount of channels/calls (doing lua IVR with > no transcoding) and conferencing. > > Any help, feedback, benchmarks or personal experience would be appreciated > > 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160216/d5115fdb/attachment-0001.html From naveen.khanna.bm at gmail.com Wed Feb 17 08:33:48 2016 From: naveen.khanna.bm at gmail.com (Naveen Khanna) Date: Wed, 17 Feb 2016 11:03:48 +0530 Subject: [Freeswitch-users] BLF of dialing party Message-ID: HI, I have come across a situation where BLF of dialing party is not lit. However, if the same user is receiving the Calls then we get proper BLF. Regards, Naveen Khanna M : +91-9911393060 | Skype : naveen.khanna.bm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160217/c3d2b904/attachment.html From jungleboogie0 at gmail.com Wed Feb 17 09:18:19 2016 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Tue, 16 Feb 2016 22:18:19 -0800 Subject: [Freeswitch-users] how to sip uri call Message-ID: Hello All, I'd like my freeswitch server to make sip uri calls to various URI's. I see this in dialplan/default.xml: If I enable this, I can call any sip URI but then I can't call an extension local to the FS instance nor out on the PSTN. I see in In the same file, I do see this: But I think there must be a method to enable URI calling for any domain--not just explicitly listed URI. If there's a confluence page that already covers this, feel free to send the article to me so I can try it out. Thanks for any assistance! -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info xmpp: jungle-boogie at jit.si From ashwinrath at gmail.com Wed Feb 17 09:20:43 2016 From: ashwinrath at gmail.com (Ashwin Rath) Date: Wed, 17 Feb 2016 11:50:43 +0530 Subject: [Freeswitch-users] How to convert a json string returned from a webservice to a json object in lua Message-ID: I am using LUA to POST to a webservice and i receive a json string . How do i convert this string to a json object ? -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160217/7ca73809/attachment.html From krice at freeswitch.org Wed Feb 17 09:26:12 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 17 Feb 2016 00:26:12 -0600 Subject: [Freeswitch-users] how to sip uri call In-Reply-To: References: Message-ID: <167301d1694c$194b20c0$4be16240$@freeswitch.org> Its all about ordering of the dialplan... first thing that matches wins... -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle Boogie Sent: Wednesday, February 17, 2016 12:18 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to sip uri call Hello All, I'd like my freeswitch server to make sip uri calls to various URI's. I see this in dialplan/default.xml: If I enable this, I can call any sip URI but then I can't call an extension local to the FS instance nor out on the PSTN. I see in In the same file, I do see this: But I think there must be a method to enable URI calling for any domain--not just explicitly listed URI. If there's a confluence page that already covers this, feel free to send the article to me so I can try it out. Thanks for any assistance! -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info xmpp: jungle-boogie at jit.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 From tarantul at gmail.com Wed Feb 17 13:03:18 2016 From: tarantul at gmail.com (Nick 'tarantul' Novikov) Date: Wed, 17 Feb 2016 13:03:18 +0300 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: Hello! Log file is large, I can't paste it in FS pastebin. You may download gzip log file here https://yadi.sk/d/pLbEHi_LoyWsS On Tue, Feb 16, 2016 at 7:11 PM, Brian West wrote: > Your logs are incomplete, need full debug logs. Please don't filter them > excessively and try on master please. > > On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli > wrote: > >> Please pastebin a complete sip trace (from fs-cli: sofia global siptrace >> on) and then put here the link to pastebin. >> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" ha >> scritto: >> >>> Hello >>> >>> I have some problem with sound after hold usage. Sometime the sound >>> disappear after callcenter agent shift unhold.For callcenter agents we use >>> sip.js (version 0.7.2). >>> In freeswitch logs I see strange lines ( >>> https://pastebin.freeswitch.org/24557). >>> Can anyone explain, what happened with audio and how to fix it? >>> >>> >>> -- >>> tarantul >>> Dios es Amor >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 > -- tarantul Dios es Amor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160217/d52e949b/attachment.html From royj at yandex.ru Wed Feb 17 14:07:16 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Wed, 17 Feb 2016 14:07:16 +0300 Subject: [Freeswitch-users] early media after bridge Message-ID: <28121455707236@web17o.yandex.ru> Hi, all Is there any chance to use the early media from the b-leg as ringback to the a-leg when a-leg is answered for play_and_get_digits for example? From cmrienzo at gmail.com Wed Feb 17 16:09:41 2016 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Wed, 17 Feb 2016 08:09:41 -0500 Subject: [Freeswitch-users] How to convert a json string returned from a webservice to a json object in lua In-Reply-To: References: Message-ID: <7837F9BB-9117-4AFA-AFBE-29D5A79A26AD@gmail.com> dkjson w/ lpeg works well for me. > On Feb 17, 2016, at 01:20, Ashwin Rath wrote: > > I am using LUA to POST to a webservice and i receive a json string . How do i convert this string to a json object ? > > -- > Ashwin Kumar Rath > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Wed Feb 17 16:27:09 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Wed, 17 Feb 2016 16:27:09 +0300 Subject: [Freeswitch-users] hiredis: realm must be defined In-Reply-To: <511661455627517@web2g.yandex.ru> References: <511661455627517@web2g.yandex.ru> Message-ID: <1070101455715629@web17o.yandex.ru> Some strange for me really. There is in hiredis.conf.xml. Also an application 'limit' with hiredis backend increments count on redis db, but after hangup does not decrease count (i look at ngrep on redis db host) Is anybody used api limit_usage with hiredis backend? 16.02.2016, 16:02, "royj at yandex.ru" : > Hi, all > Can anybody clarify what mean 'hiredis: realm must be defined'? Where is the misconfiguration? > I am trying to use hiredis backend for limits and call api like 'api limit_usage hiredis myrealm id', got api/response '-1' and FreeSWITCH says in log 'hiredis: realm must be defined'. > Module mod_hiredis loaded and exists. > For hash backend works fine. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 asilva at wirelessmundi.com Wed Feb 17 16:31:28 2016 From: asilva at wirelessmundi.com (Antonio Silva) Date: Wed, 17 Feb 2016 14:31:28 +0100 Subject: [Freeswitch-users] Disable PRACK messages Message-ID: <56C47630.7070402@wirelessmundi.com> Hi, Is it possible to disable or not to send prack messages? I've set in the sip profile the option enable-100rel=false, but FS still sends the prack message... My problem is that after receive the 180 from the remote side, FS sends the prack message and the remote side reply with SIP/2.0 481 Call/Transaction Does Not Exist, This make FS to cancel the call. I know that this is an issue on the remote side... but i want to avoid this message if possible. Full sip trace in https://pastebin.freeswitch.org/24562 Thanks, Ant?nio From brian at freeswitch.org Wed Feb 17 17:31:11 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Feb 2016 08:31:11 -0600 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: bug reports go on JIRA please. On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov wrote: > Hello! > > Log file is large, I can't paste it in FS pastebin. > You may download gzip log file here > https://yadi.sk/d/pLbEHi_LoyWsS > > On Tue, Feb 16, 2016 at 7:11 PM, Brian West wrote: > >> Your logs are incomplete, need full debug logs. Please don't filter them >> excessively and try on master please. >> >> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli >> wrote: >> >>> Please pastebin a complete sip trace (from fs-cli: sofia global siptrace >>> on) and then put here the link to pastebin. >>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" ha >>> scritto: >>> >>>> Hello >>>> >>>> I have some problem with sound after hold usage. Sometime the sound >>>> disappear after callcenter agent shift unhold.For callcenter agents we use >>>> sip.js (version 0.7.2). >>>> In freeswitch logs I see strange lines ( >>>> https://pastebin.freeswitch.org/24557). >>>> Can anyone explain, what happened with audio and how to fix it? >>>> >>>> >>>> -- >>>> tarantul >>>> Dios es Amor >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >> > > > > -- > tarantul > Dios es Amor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160217/0c3480d0/attachment.html From brian at freeswitch.org Wed Feb 17 17:31:42 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Feb 2016 08:31:42 -0600 Subject: [Freeswitch-users] early media after bridge In-Reply-To: <28121455707236@web17o.yandex.ru> References: <28121455707236@web17o.yandex.ru> Message-ID: It SHOULD already be doing this unless you're ignoring early media on that oubound leg. On Wed, Feb 17, 2016 at 5:07 AM, wrote: > Hi, all > Is there any chance to use the early media from the b-leg as ringback to > the a-leg when a-leg is answered for play_and_get_digits for example? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160217/a7cfc111/attachment-0001.html From royj at yandex.ru Wed Feb 17 17:57:19 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Wed, 17 Feb 2016 17:57:19 +0300 Subject: [Freeswitch-users] early media after bridge In-Reply-To: References: <28121455707236@web17o.yandex.ru> Message-ID: <1543381455721039@web22g.yandex.ru> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160217/c200098e/attachment.html From tarantul at gmail.com Wed Feb 17 23:32:37 2016 From: tarantul at gmail.com (Nick 'tarantul' Novikov) Date: Wed, 17 Feb 2016 23:32:37 +0300 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: Hello! Done. https://freeswitch.org/jira/browse/FS-8840 On Wed, Feb 17, 2016 at 5:31 PM, Brian West wrote: > bug reports go on JIRA please. > > On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov < > tarantul at gmail.com> wrote: > >> Hello! >> >> Log file is large, I can't paste it in FS pastebin. >> You may download gzip log file here >> https://yadi.sk/d/pLbEHi_LoyWsS >> >> On Tue, Feb 16, 2016 at 7:11 PM, Brian West wrote: >> >>> Your logs are incomplete, need full debug logs. Please don't filter >>> them excessively and try on master please. >>> >>> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli >> > wrote: >>> >>>> Please pastebin a complete sip trace (from fs-cli: sofia global >>>> siptrace on) and then put here the link to pastebin. >>>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" >>>> ha scritto: >>>> >>>>> Hello >>>>> >>>>> I have some problem with sound after hold usage. Sometime the sound >>>>> disappear after callcenter agent shift unhold.For callcenter agents we use >>>>> sip.js (version 0.7.2). >>>>> In freeswitch logs I see strange lines ( >>>>> https://pastebin.freeswitch.org/24557). >>>>> Can anyone explain, what happened with audio and how to fix it? >>>>> >>>>> >>>>> -- >>>>> tarantul >>>>> Dios es Amor >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>> >> >> >> >> -- >> tarantul >> Dios es Amor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- tarantul Dios es Amor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160217/07e68e75/attachment.html From DEdwards at vertical.com Thu Feb 18 00:16:47 2016 From: DEdwards at vertical.com (Dan Edwards) Date: Wed, 17 Feb 2016 21:16:47 +0000 Subject: [Freeswitch-users] Can FS offer both internal & external IP address? Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com> I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario. Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change. If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. Is there a way to offer both and allow ICE on the client side to figure out which one to use? Thanks, Dan From brian at freeswitch.org Thu Feb 18 00:34:04 2016 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Feb 2016 15:34:04 -0600 Subject: [Freeswitch-users] Can FS offer both internal & external IP address? In-Reply-To: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com> References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com> Message-ID: Actually no, what you do in this case is set the local-network-acl, it will then know which one to use based on the request. On Wed, Feb 17, 2016 at 3:16 PM, Dan Edwards wrote: > I'm running FS behind Nginx, so I explicitly removed the local IP address > from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside > users work fine in that scenario. > > Internal users also work, as long as we turn on NAT reflection in our > router. I'm looking for a way to not require that router change. > > If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as > candidates in the initial INVITE, the browser could then use ICE to > determine which IP address was correct. > > Is there a way to offer both and allow ICE on the client side to figure > out which one to use? > > Thanks, > 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 > -- *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/20160217/660fc1f4/attachment.html From DEdwards at vertical.com Thu Feb 18 00:45:05 2016 From: DEdwards at vertical.com (Dan Edwards) Date: Wed, 17 Feb 2016 21:45:05 +0000 Subject: [Freeswitch-users] Can FS offer both internal & external IP address? In-Reply-To: References: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA45@PHXEX2.vertical.com> Message-ID: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997FA68@PHXEX2.vertical.com> ?based on the request? is what?s actually causing the problem. Since the requests come through a proxy, every ?remote_ip? is always the local IP, so FS cannot make a determination based on that. I added a deny to the local IP which caused FS to offer the external IP and we can make that work if our customers turn on the NAT reflection. As a test, I modified switch_core_media.c:generate_m to throw in the local IP and that did fix our issue. I was hoping for a configuration option. Thanks, Dan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, February 17, 2016 4:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Can FS offer both internal & external IP address? Actually no, what you do in this case is set the local-network-acl, it will then know which one to use based on the request. On Wed, Feb 17, 2016 at 3:16 PM, Dan Edwards > wrote: I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario. Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change. If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. Is there a way to offer both and allow ICE on the client side to figure out which one to use? Thanks, 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 -- Brian West brian at freeswitch.org [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160217/f195417d/attachment.html From ahabiba at gmail.com Thu Feb 18 00:54:36 2016 From: ahabiba at gmail.com (Ahmed Habiba) Date: Thu, 18 Feb 2016 00:54:36 +0300 Subject: [Freeswitch-users] Can FS offer both internal & external IP address? In-Reply-To: References: Message-ID: <59068F3A-AFE8-463F-AE57-7E91E10A673A@gmail.com> It is all about how you want things to run, FS allow to have multiple profile each of them can work on different ip considering that all users can call each other even they are registered on different IPs here is below and example: profile1: Name?> Internal IP ?> 192.168.1.1 SIP Port ?> 5060 Sip_profile.xml: However of the above configuration still both internal and external will be able to call each other based on the below dial string , the ?*/? assure that users from different profile can call each other conf/directory/default.xml : > On Feb 18, 2016, at 12:17 AM, freeswitch-users-request at lists.freeswitch.org wrote: > > From: Dan Edwards > > Subject: [Freeswitch-users] Can FS offer both internal & external IP address? > Date: February 18, 2016 at 12:16:47 AM GMT+3 > To: "freeswitch-users at lists.freeswitch.org " > > Reply-To: FreeSWITCH Users Help > > > > I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario. > > Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change. > > If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. > > Is there a way to offer both and allow ICE on the client side to figure out which one to use? > > Thanks, > Dan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/6561d190/attachment-0001.html From jprangi at gmail.com Thu Feb 18 01:01:30 2016 From: jprangi at gmail.com (Jai Rangi) Date: Wed, 17 Feb 2016 14:01:30 -0800 Subject: [Freeswitch-users] Oneway audio issues in freeswitch In-Reply-To: <67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com> References: <67F80D46-E511-459C-A72B-E966671D4AB3@dchorton.com> Message-ID: Hello Dave, Sorry for the delay, have been out of office. Attached the the debug log file, with siptrace on. This is coming from default configuration. I can confirm the issue happens only when this line is commented out. Not sure if this qualifies for bug, I can open a ticket in Jira. Thank you, -Jai On Sat, Feb 13, 2016 at 7:49 PM, Dave Horton wrote: > Redo with sofia global siptrace on so we can see the SIP messaging > On Feb 13, 2016, at 12:45 PM, 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160217/42e3e0fe/attachment-0001.html -------------- next part -------------- 2016-02-17 13:47:25.637398 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv 2016-02-17 13:47:25.637398 [NOTICE] mod_logfile.c:213 New log started. recv 864 bytes from udp/[66.166.9.194]:39164 at 13:47:27.794790: ------------------------------------------------------------------------ INVITE sip:1001 at 216.209.211.222 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport From: ;tag=243846933 To: Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 40 INVITE Contact: Max-Forwards: 70 User-Agent: Grandstream Wave/IOS 1.0.19 Privacy: none P-Preferred-Identity: 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: 230 v=0 o=1000 8000 8000 IN IP4 10.0.1.12 s=SIP Call c=IN IP4 10.0.1.12 t=0 0 m=audio 10270 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 ------------------------------------------------------------------------ send 310 bytes to udp/[66.166.9.194]:39164 at 13:47:27.795559: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 40 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Content-Length: 0 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1000 at 216.209.211.222 [4008c053-8780-4987-b84f-010b67013bda] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_NEW 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.777377 [DEBUG] sofia.c:9248 sofia/internal/1000 at 216.209.211.222 receiving invite from 66.166.9.194:39164 version: 1.6.6 64bit 2016-02-17 13:47:27.777377 [DEBUG] sofia.c:9415 IP 66.166.9.194 Rejected by acl "domains". Falling back to Digest auth. 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.797388 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/1000 at 216.209.211.222) State NEW send 814 bytes to udp/[66.166.9.194]:39164 at 13:47:27.798293: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: ;tag=Drpp4yp9j1X2m Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 40 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, 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="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2016-02-17 13:47:27.797388 [DEBUG] sofia.c:2147 detaching session 4008c053-8780-4987-b84f-010b67013bda recv 267 bytes from udp/[66.166.9.194]:39164 at 13:47:27.843925: ------------------------------------------------------------------------ ACK sip:1001 at 216.209.211.222 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK281160191;rport From: ;tag=243846933 To: ;tag=Drpp4yp9j1X2m Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 40 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1111 bytes from udp/[66.166.9.194]:39164 at 13:47:27.860855: ------------------------------------------------------------------------ INVITE sip:1001 at 216.209.211.222 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport From: ;tag=243846933 To: Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 INVITE Contact: Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006 Max-Forwards: 70 User-Agent: Grandstream Wave/IOS 1.0.19 Privacy: none P-Preferred-Identity: 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: 230 v=0 o=1000 8000 8000 IN IP4 10.0.1.12 s=SIP Call c=IN IP4 10.0.1.12 t=0 0 m=audio 10270 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 ------------------------------------------------------------------------ send 311 bytes to udp/[66.166.9.194]:39164 at 13:47:27.862041: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Content-Length: 0 ------------------------------------------------------------------------ 2016-02-17 13:47:27.857368 [DEBUG] sofia.c:2255 Re-attaching to session 4008c053-8780-4987-b84f-010b67013bda 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:9248 sofia/internal/1000 at 216.209.211.222 receiving invite from 66.166.9.194:39164 version: 1.6.6 64bit 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:9415 IP 66.166.9.194 Rejected by acl "domains". Falling back to Digest auth. 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:10549 Setting NAT mode based on nat.auto 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [received][100] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:6770 Remote SDP: 4008c053-8780-4987-b84f-010b67013bda v=0 4008c053-8780-4987-b84f-010b67013bda o=1000 8000 8000 IN IP4 10.0.1.12 4008c053-8780-4987-b84f-010b67013bda s=SIP Call 4008c053-8780-4987-b84f-010b67013bda c=IN IP4 10.0.1.12 4008c053-8780-4987-b84f-010b67013bda t=0 0 4008c053-8780-4987-b84f-010b67013bda m=audio 10270 RTP/AVP 0 8 101 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:8 PCMA/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000 4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-15 4008c053-8780-4987-b84f-010b67013bda a=ptime:20 4008c053-8780-4987-b84f-010b67013bda 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] sofia.c:7125 (sofia/internal/1000 at 216.209.211.222) State Change CS_NEW -> CS_INIT 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_INIT 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1000 at 216.209.211.222) State INIT 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:88 sofia/internal/1000 at 216.209.211.222 SOFIA INIT 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1000 at 216.209.211.222 Standard INIT 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1000 at 216.209.211.222) State Change CS_INIT -> CS_ROUTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1000 at 216.209.211.222) State INIT going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_ROUTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:2247 (sofia/internal/1000 at 216.209.211.222) Callstate Change DOWN -> RINGING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1000 at 216.209.211.222) State ROUTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:141 sofia/internal/1000 at 216.209.211.222 SOFIA ROUTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1000 at 216.209.211.222 Standard ROUTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->unloop] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->tod_example] continue=true 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Date/Time Match (PASS) [tod_example] break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(open=true) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->holiday_example] continue=true 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Date/TimeMatch (FAIL) [holiday_example] break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->global-intercept] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->group-intercept] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->intercept-ext] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->redial] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->global] continue=true 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${default_password}(qwerty at 123$) =~ /^1234$/ break=never 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [global] ${switch_r_sdp}(v=0 4008c053-8780-4987-b84f-010b67013bda o=1000 8000 8000 IN IP4 10.0.1.12 4008c053-8780-4987-b84f-010b67013bda s=SIP Call 4008c053-8780-4987-b84f-010b67013bda c=IN IP4 10.0.1.12 4008c053-8780-4987-b84f-010b67013bda t=0 0 4008c053-8780-4987-b84f-010b67013bda m=audio 10270 RTP/AVP 0 8 101 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:8 PCMA/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000 4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-15 4008c053-8780-4987-b84f-010b67013bda a=ptime:20 4008c053-8780-4987-b84f-010b67013bda ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Absolute Condition [global] 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->snom-demo-2] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->snom-demo-1] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->eavesdrop] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->eavesdrop] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call_return] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->del-group] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->add-group] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call-group-simo] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->call-group-order] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->extension-intercom] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 parsing [default->Local_Extension] continue=false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action export(dialed_extension=1001) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(1 b s execute_extension::dx XML features) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(3 b s execute_extension::cf XML features) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(ringback=${us-ring}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(transfer_ringback=local_stream://moh) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(call_timeout=30) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(hangup_after_bridge=true) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(continue_on_fail=true) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bridge(user/${dialed_extension}@${domain_name}) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action answer() 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action sleep(1000) 4008c053-8780-4987-b84f-010b67013bda Dialplan: sofia/internal/1000 at 216.209.211.222 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1000 at 216.209.211.222) State Change CS_ROUTING -> CS_EXECUTE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1000 at 216.209.211.222) State ROUTING going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_EXECUTE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1000 at 216.209.211.222) State EXECUTE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_sofia.c:196 sofia/internal/1000 at 216.209.211.222 SOFIA EXECUTE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1000 at 216.209.211.222 Standard EXECUTE 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(open=true) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [open]=[true] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-spymap/1000/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/1000/1001) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/global/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 export(RFC2822_DATE=Wed, 17 Feb 2016 13:47:27 -0800) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 export(dialed_extension=1001) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [dialed_extension]=[1001] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(1 b s execute_extension::dx XML features) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *1 execute_extension::dx XML features 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(2 b s record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/1000.2016-02-17-13-47-27.wav) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::/usr/local/freeswitch/var/lib/freeswitch/recordings/1000.2016-02-17-13-47-27.wav 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(3 b s execute_extension::cf XML features) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *3 execute_extension::cf XML features 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bind_meta_app(4 b s execute_extension::att_xfer XML features) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *4 execute_extension::att_xfer XML features 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(ringback=%(2000,4000,440,480)) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [ringback]=[%(2000,4000,440,480)] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(transfer_ringback=local_stream://moh) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [transfer_ringback]=[local_stream://moh] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(call_timeout=30) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [call_timeout]=[30] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(hangup_after_bridge=true) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.877397 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [hangup_after_bridge]=[true] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(continue_on_fail=true) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [continue_on_fail]=[true] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-call_return/1001/1000) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/1001/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 set(called_party_callgroup=techsupport) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] mod_dptools.c:1498 SET sofia/internal/1000 at 216.209.211.222 [called_party_callgroup]=[techsupport] 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/techsupport/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial_ext/global/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 hash(insert/216.209.211.222-last_dial/techsupport/4008c053-8780-4987-b84f-010b67013bda) 4008c053-8780-4987-b84f-010b67013bda EXECUTE sofia/internal/1000 at 216.209.211.222 bridge(user/1001 at 216.209.211.222) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800] to event 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [dialed_extension]=[1001] to event 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 17 Feb 2016 13:47:27 -0800] to event 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_channel.c:1247 sofia/internal/1000 at 216.209.211.222 EXPORTING[export_vars] [dialed_extension]=[1001] to event 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/1001 at 10.0.1.2:40886 [15df2523-1e92-49ca-bf21-bbf07d51c29e] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:4776 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_NEW -> CS_INIT 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_INIT 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1001 at 10.0.1.2:40886) State INIT 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:88 sofia/internal/1001 at 10.0.1.2:40886 SOFIA INIT 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia_glue.c:1228 sip:1001 at 66.166.9.194:41958 Setting proxy route to sofia/internal/1001 at 10.0.1.2:40886 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia_glue.c:1257 sofia/internal/1001 at 10.0.1.2:40886 sending invite version: 1.6.6 64bit 15df2523-1e92-49ca-bf21-bbf07d51c29e Local SDP: 15df2523-1e92-49ca-bf21-bbf07d51c29e v=0 15df2523-1e92-49ca-bf21-bbf07d51c29e o=FreeSWITCH 1455713039 1455713040 IN IP4 216.209.211.222 15df2523-1e92-49ca-bf21-bbf07d51c29e s=FreeSWITCH 15df2523-1e92-49ca-bf21-bbf07d51c29e c=IN IP4 216.209.211.222 15df2523-1e92-49ca-bf21-bbf07d51c29e t=0 0 15df2523-1e92-49ca-bf21-bbf07d51c29e m=audio 32608 RTP/AVP 0 8 101 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:0 PCMU/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:8 PCMA/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:101 telephone-event/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=fmtp:101 0-16 15df2523-1e92-49ca-bf21-bbf07d51c29e a=ptime:20 15df2523-1e92-49ca-bf21-bbf07d51c29e a=sendrecv 15df2523-1e92-49ca-bf21-bbf07d51c29e 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at 10.0.1.2:40886 Standard INIT 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_INIT -> CS_ROUTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/1001 at 10.0.1.2:40886) State INIT going to sleep 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_ROUTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1001 at 10.0.1.2:40886) State ROUTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] mod_sofia.c:141 sofia/internal/1001 at 10.0.1.2:40886 SOFIA ROUTING send 1185 bytes to udp/[66.166.9.194]:41958 at 13:47:27.912034: ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_ROUTING -> CS_CONSUME_MEDIA INVITE sip:1001 at 10.0.1.2:40886 SIP/2.0 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1001 at 10.0.1.2:40886) State ROUTING going to sleep Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK38Xat6Z71UHUN Route: Max-Forwards: 69 From: "Extension 1000" ;tag=Fa977mrgDKa8B To: Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536759 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, 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 Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info P-Asserted-Identity: "Extension 1000" v=0 o=FreeSWITCH 1455713039 1455713040 IN IP4 216.209.211.222 s=FreeSWITCH c=IN IP4 216.209.211.222 t=0 0 m=audio 32608 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_CONSUME_MEDIA 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [calling][0] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1001 at 10.0.1.2:40886) State CONSUME_MEDIA 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:27.897386 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/1001 at 10.0.1.2:40886) State CONSUME_MEDIA going to sleep recv 450 bytes from udp/[66.166.9.194]:41958 at 13:47:28.028660: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN From: "Extension 1000" ;tag=Fa977mrgDKa8B To: Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536759 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 ------------------------------------------------------------------------ recv 535 bytes from udp/[66.166.9.194]:41958 at 13:47:28.036530: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN From: "Extension 1000" ;tag=Fa977mrgDKa8B To: ;tag=2072085022 Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536759 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 ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [proceeding][180] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [NOTICE] sofia.c:6862 Ring-Ready sofia/internal/1001 at 10.0.1.2:40886! 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:28.037389 [DEBUG] switch_channel.c:3340 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change DOWN -> RINGING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [INFO] switch_ivr_originate.c:1216 Sending early media 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1000 at 216.209.211.222 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at 216.209.211.222 Original read codec set to PCMU:0 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:4485 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf send payload to 101 recv payload to 101 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1000 at 216.209.211.222] 216.209.211.222 port 25166 -> 10.0.1.12 port 10270 codec: 0 ms: 20 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6332 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf send payload to 101 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6339 sofia/internal/1000 at 216.209.211.222 Set 2833 dtmf receive payload to 101 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [DEBUG] switch_core_media.c:6362 sofia/internal/1000 at 216.209.211.222 Set rtp dtmf delay to 40 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.037389 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at 216.209.211.222! 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_channel.c:3468 (sofia/internal/1000 at 216.209.211.222) Callstate Change RINGING -> EARLY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] mod_sofia.c:2330 Ring SDP: 4008c053-8780-4987-b84f-010b67013bda v=0 4008c053-8780-4987-b84f-010b67013bda o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222 4008c053-8780-4987-b84f-010b67013bda s=FreeSWITCH 4008c053-8780-4987-b84f-010b67013bda c=IN IP4 216.209.211.222 4008c053-8780-4987-b84f-010b67013bda t=0 0 4008c053-8780-4987-b84f-010b67013bda m=audio 25166 RTP/AVP 0 101 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000 4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-16 4008c053-8780-4987-b84f-010b67013bda a=ptime:20 4008c053-8780-4987-b84f-010b67013bda a=sendrecv 4008c053-8780-4987-b84f-010b67013bda 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_core_codec.c:221 sofia/internal/1000 at 216.209.211.222 Push codec L16:100 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.057400 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] send 1072 bytes to udp/[66.166.9.194]:39164 at 13:47:28.060325: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, 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: "1001" v=0 o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222 s=FreeSWITCH c=IN IP4 216.209.211.222 t=0 0 m=audio 25166 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.077397 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [early][183] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:28.817431 [INFO] switch_rtp.c:6616 Auto Changing audio port from 10.0.1.12:10270 to 66.166.9.194:38124 recv 767 bytes from udp/[66.166.9.194]:41958 at 13:47:30.718270: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK38Xat6Z71UHUN From: "Extension 1000" ;tag=Fa977mrgDKa8B To: ;tag=2072085022 Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536759 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: 228 v=0 o=1001 8001 8000 IN IP4 10.0.1.2 s=SIP Call c=IN IP4 10.0.1.2 t=0 0 m=audio 35544 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 ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [completing][200] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6770 Remote SDP: 15df2523-1e92-49ca-bf21-bbf07d51c29e v=0 15df2523-1e92-49ca-bf21-bbf07d51c29e o=1001 8001 8000 IN IP4 10.0.1.2 15df2523-1e92-49ca-bf21-bbf07d51c29e s=SIP Call 15df2523-1e92-49ca-bf21-bbf07d51c29e c=IN IP4 10.0.1.2 15df2523-1e92-49ca-bf21-bbf07d51c29e t=0 0 15df2523-1e92-49ca-bf21-bbf07d51c29e m=audio 35544 RTP/AVP 0 8 101 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:0 PCMU/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:8 PCMA/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=rtpmap:101 telephone-event/8000 15df2523-1e92-49ca-bf21-bbf07d51c29e a=fmtp:101 0-15 15df2523-1e92-49ca-bf21-bbf07d51c29e a=ptime:20 15df2523-1e92-49ca-bf21-bbf07d51c29e send 367 bytes to udp/[66.166.9.194]:41958 at 13:47:30.723049: ------------------------------------------------------------------------ ACK sip:1001 at 10.0.1.2:40886 SIP/2.0 Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK4HQ3U1gBZ47DH Max-Forwards: 70 From: "Extension 1000" ;tag=Fa977mrgDKa8B To: ;tag=2072085022 Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536759 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] sofia.c:6760 Channel sofia/internal/1001 at 10.0.1.2:40886 entering state [ready][200] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4077 Set telephone-event payload to 101 at 8000 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/1001 at 10.0.1.2:40886 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at 10.0.1.2:40886 Original read codec set to PCMU:0 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4429 Set telephone-event payload to 101 at 8000 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:4485 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf send payload to 101 recv payload to 101 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6033 AUDIO RTP [sofia/internal/1001 at 10.0.1.2:40886] 216.209.211.222 port 32608 -> 10.0.1.2 port 35544 codec: 0 ms: 20 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_rtp.c:3802 Starting timer [soft] 160 bytes per 20ms 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:1939 Setting Jitterbuffer to 60ms (3 frames) (50 max frames) 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6332 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf send payload to 101 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6339 sofia/internal/1001 at 10.0.1.2:40886 Set 2833 dtmf receive payload to 101 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_core_media.c:6362 sofia/internal/1001 at 10.0.1.2:40886 Set rtp dtmf delay to 40 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [NOTICE] sofia.c:7724 Channel [sofia/internal/1001 at 10.0.1.2:40886] has been answered 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.717395 [DEBUG] switch_channel.c:3767 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change RINGING -> ACTIVE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_core_codec.c:246 sofia/internal/1000 at 216.209.211.222 Restore previous codec PCMU:0. 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] mod_sofia.c:799 Local SDP sofia/internal/1000 at 216.209.211.222: 4008c053-8780-4987-b84f-010b67013bda v=0 4008c053-8780-4987-b84f-010b67013bda o=FreeSWITCH 1455720482 1455720484 IN IP4 216.209.211.222 4008c053-8780-4987-b84f-010b67013bda s=FreeSWITCH 4008c053-8780-4987-b84f-010b67013bda c=IN IP4 216.209.211.222 4008c053-8780-4987-b84f-010b67013bda t=0 0 4008c053-8780-4987-b84f-010b67013bda m=audio 25166 RTP/AVP 0 101 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:0 PCMU/8000 4008c053-8780-4987-b84f-010b67013bda a=rtpmap:101 telephone-event/8000 4008c053-8780-4987-b84f-010b67013bda a=fmtp:101 0-16 4008c053-8780-4987-b84f-010b67013bda a=ptime:20 4008c053-8780-4987-b84f-010b67013bda a=sendrecv 4008c053-8780-4987-b84f-010b67013bda 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [NOTICE] switch_ivr_originate.c:3550 Channel [sofia/internal/1000 at 216.209.211.222] has been answered send 1094 bytes to udp/[66.166.9.194]:39164 at 13:47:30.743468: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, 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 Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 222 P-Asserted-Identity: "Outbound Call" v=0 o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222 s=FreeSWITCH c=IN IP4 216.209.211.222 t=0 0 m=audio 25166 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_channel.c:3767 (sofia/internal/1000 at 216.209.211.222) Callstate Change EARLY -> ACTIVE 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [completed][200] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1001 at 10.0.1.2:40886] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_originate.c:3608 Originate Resulted in Success: [sofia/internal/1001 at 10.0.1.2:40886] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_media.c:9156 sofia/internal/1001 at 10.0.1.2:40886 PAUSE Jitterbuffer 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.737383 [DEBUG] switch_core_media.c:9156 sofia/internal/1000 at 216.209.211.222 PAUSE Jitterbuffer 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_ivr_bridge.c:1591 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_EXCHANGE_MEDIA 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1001 at 10.0.1.2:40886) State EXCHANGE_MEDIA 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:30.737383 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:30.757403 [DEBUG] switch_rtp.c:6654 Correct audio ip/port confirmed. send 1094 bytes to udp/[66.166.9.194]:39164 at 13:47:31.244642: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1578402832;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, 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 Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 222 P-Asserted-Identity: "Outbound Call" v=0 o=FreeSWITCH 1455720482 1455720483 IN IP4 216.209.211.222 s=FreeSWITCH c=IN IP4 216.209.211.222 t=0 0 m=audio 25166 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 764 bytes from udp/[66.166.9.194]:39164 at 13:47:31.321742: ------------------------------------------------------------------------ ACK sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK145924562;rport From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 ACK Contact: Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006 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 ------------------------------------------------------------------------ recv 764 bytes from udp/[66.166.9.194]:39164 at 13:47:31.327208: ------------------------------------------------------------------------ ACK sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK145924562;rport From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 41 ACK Contact: Proxy-Authorization: Digest username="1000", realm="216.209.211.222", nonce="696ccc87-c214-4a8a-870f-7e3d69ca016d", uri="sip:1001 at 216.209.211.222", response="73ea6df6242e1f2e5cc6e4d6bad9be3e", algorithm=MD5, cnonce="15059615", qop=auth, nc=00000006 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 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:31.317425 [DEBUG] sofia.c:6760 Channel sofia/internal/1000 at 216.209.211.222 entering state [ready][200] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:31.517381 [INFO] switch_rtp.c:6616 Auto Changing audio port from 10.0.1.2:35544 to 66.166.9.194:46630 recv 519 bytes from udp/[66.166.9.194]:39164 at 13:47:37.588043: ------------------------------------------------------------------------ BYE sip:1001 at 216.209.211.222:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1092657280;rport From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 42 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 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] sofia.c:952 Hangup sofia/internal/1000 at 216.209.211.222 [CS_EXECUTE] [NORMAL_CLEARING] send 466 bytes to udp/[66.166.9.194]:39164 at 13:47:37.598615: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.12:40886;branch=z9hG4bK1092657280;rport=39164;received=66.166.9.194 From: ;tag=243846933 To: ;tag=e1FF6S7cgamNg Call-ID: 930817686-40886-5 at BA.A.B.BC CSeq: 42 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1000 at 216.209.211.222] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:699 sofia/internal/1000 at 216.209.211.222 ending bridge by request from write function 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD DONE [sofia/internal/1001 at 10.0.1.2:40886] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_ivr_bridge.c:881 Hangup sofia/internal/1001 at 10.0.1.2:40886 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:542 (sofia/internal/1001 at 10.0.1.2:40886) State EXCHANGE_MEDIA going to sleep 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_HANGUP 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1001 at 10.0.1.2:40886) Callstate Change ACTIVE -> HANGUP 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:1689 sofia/internal/1001 at 10.0.1.2:40886 skip receive message [UNBRIDGE] (channel is hungup already) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_ivr_bridge.c:1692 sofia/internal/1000 at 216.209.211.222 skip receive message [UNBRIDGE] (channel is hungup already) 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001 at 10.0.1.2:40886) State HANGUP 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:425 sofia/internal/1001 at 10.0.1.2:40886 Overriding SIP cause 480 with 200 from the other leg 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1001 at 10.0.1.2:40886 hanging up, cause: NORMAL_CLEARING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:2796 sofia/internal/1000 at 216.209.211.222 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1000 at 216.209.211.222) State EXECUTE going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_HANGUP 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1000 at 216.209.211.222) Callstate Change ACTIVE -> HANGUP 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1000 at 216.209.211.222) State HANGUP 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1000 at 216.209.211.222 hanging up, cause: NORMAL_CLEARING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:484 Sending BYE to sofia/internal/1001 at 10.0.1.2:40886 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at 10.0.1.2:40886 Standard HANGUP, cause: NORMAL_CLEARING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001 at 10.0.1.2:40886) State HANGUP going to sleep 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_HANGUP -> CS_REPORTING send 559 bytes to udp/[66.166.9.194]:41958 at 13:47:37.609113: ------------------------------------------------------------------------ BYE sip:1001 at 10.0.1.2:40886 SIP/2.0 Via: SIP/2.0/UDP 216.209.211.222;rport;branch=z9hG4bK5tgvXv1evDy0c Max-Forwards: 70 From: "Extension 1000" ;tag=Fa977mrgDKa8B To: ;tag=2072085022 Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536760 BYE15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_REPORTING User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001 at 10.0.1.2:40886) State REPORTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001 at 10.0.1.2:40886 Standard REPORTING, cause: NORMAL_CLEARING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001 at 10.0.1.2:40886) State REPORTING going to sleep 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1001 at 10.0.1.2:40886) State Change CS_REPORTING -> CS_DESTROY 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:1646 Session 7 (sofia/internal/1001 at 10.0.1.2:40886) Locked, Waiting on external entities 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1664 Session 7 (sofia/internal/1001 at 10.0.1.2:40886) Ended 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1001 at 10.0.1.2:40886 [CS_DESTROY] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 216.209.211.222 Standard HANGUP, cause: NORMAL_CLEARING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1000 at 216.209.211.222) State HANGUP going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1000 at 216.209.211.222) State Change CS_HANGUP -> CS_REPORTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1001 at 10.0.1.2:40886) Running State Change CS_DESTROY 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001 at 10.0.1.2:40886) State DESTROY 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:341 sofia/internal/1001 at 10.0.1.2:40886 SOFIA DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_REPORTING 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001 at 10.0.1.2:40886 Standard DESTROY 15df2523-1e92-49ca-bf21-bbf07d51c29e 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001 at 10.0.1.2:40886) State DESTROY going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1000 at 216.209.211.222) State REPORTING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1000 at 216.209.211.222 Standard REPORTING, cause: NORMAL_CLEARING 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1000 at 216.209.211.222) State REPORTING going to sleep 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1000 at 216.209.211.222) State Change CS_REPORTING -> CS_DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_session.c:1646 Session 6 (sofia/internal/1000 at 216.209.211.222) Locked, Waiting on external entities 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1664 Session 6 (sofia/internal/1000 at 216.209.211.222) Ended 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/1000 at 216.209.211.222 [CS_DESTROY] 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1000 at 216.209.211.222) Running State Change CS_DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1000 at 216.209.211.222) State DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] mod_sofia.c:341 sofia/internal/1000 at 216.209.211.222 SOFIA DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1000 at 216.209.211.222 Standard DESTROY 4008c053-8780-4987-b84f-010b67013bda 2016-02-17 13:47:37.597418 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1000 at 216.209.211.222) State DESTROY going to sleep recv 501 bytes from udp/[66.166.9.194]:41958 at 13:47:37.651003: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 216.209.211.222;rport=5060;branch=z9hG4bK5tgvXv1evDy0c From: "Extension 1000" ;tag=Fa977mrgDKa8B To: ;tag=2072085022 Call-ID: e124e149-5062-1234-9bb4-c6827b410b7a CSeq: 87536760 BYE 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-Length: 0 ------------------------------------------------------------------------ 2016-02-17 13:47:41.537380 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/1000.csv 2016-02-17 13:47:41.537380 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/var/log/freeswitch/cdr-csv/Master.csv From Alexander.Haugg at c4b.de Thu Feb 18 11:29:38 2016 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 18 Feb 2016 08:29:38 +0000 Subject: [Freeswitch-users] Web RTC -> No Candidate found Message-ID: Hi, I have the problem, that the freeswitch don't accept the network internal candidate. The local IP address of the fs is 192.168.241.5 The local IP address of the webrtc Client is 192.168.241.2 In the acl.conf.xml is set: In the sofia profile is set: The following SDP is send by the client: v=0 o=- 222075295642546336 815544600 IN IP4 127.0.0.1 s=IceLink t=0 0 m=audio 30450 RTP/SAVPF 0 8 c=IN IP4 192.168.62.1 a=rtcp:30450 IN IP4 192.168.62.1 a=ice-ufrag:532479e3 a=ice-pwd:4b066f2e64633d218085fc39c49a443c a=sendrecv a=rtcp-mux a=fingerprint:sha-256 FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36 a=setup:actpass a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ssrc:967296980 cname:c2088983 a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 192.168.241.2 44376 typ host a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 192.168.241.2 44376 typ host The output on the fs_cli: 2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 Original read codec set to PCMU:0 2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate ACL defined, Defaulting to wan.auto 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtcp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 no suitable candidates found. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/e1b90daf/attachment.html From brian at freeswitch.org Thu Feb 18 17:31:25 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Feb 2016 08:31:25 -0600 Subject: [Freeswitch-users] Web RTC -> No Candidate found In-Reply-To: References: Message-ID: Try apply-candidate-acl On Thu, Feb 18, 2016 at 2:29 AM, Alexander Haugg wrote: > Hi, > > > > I have the problem, that the freeswitch don?t accept the network internal > candidate. > > > > The local IP address of the fs is 192.168.241.5 > > The local IP address of the webrtc Client is 192.168.241.2 > > > > In the acl.conf.xml is set: > > > > > > > > > > In the sofia profile is set: > > > > > > > > The following SDP is send by the client: > > v=0 > > o=- 222075295642546336 815544600 IN IP4 127.0.0.1 > > s=IceLink > > t=0 0 > > m=audio 30450 RTP/SAVPF 0 8 > > c=IN IP4 192.168.62.1 > > a=rtcp:30450 IN IP4 192.168.62.1 > > a=ice-ufrag:532479e3 > > a=ice-pwd:4b066f2e64633d218085fc39c49a443c > > a=sendrecv > > a=rtcp-mux > > a=fingerprint:sha-256 > FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36 > > a=setup:actpass > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=ssrc:967296980 cname:c2088983 > > a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 > 192.168.241.2 44376 typ host > > a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 > 192.168.241.2 44376 typ host > > > > The output on the fs_cli: > > 2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 > (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec > Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec > sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 PCMU/8000 20 ms 160 samples > 64000 bits 1 channels > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 > sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 Original read codec set to > PCMU:0 > > 2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate > ACL defined, Defaulting to wan.auto > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio > Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376 > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio > Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760 > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for > rtp candidate. > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for > rtcp candidate. > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 > sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 no suitable candidates found. > > 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in > SDP. Disable 2833 dtmf and switch to INFO > > 2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup > sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION > > > > Thanks a lot! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160218/0cf8aafb/attachment-0001.html From mike at jerris.com Thu Feb 18 19:28:02 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 Feb 2016 11:28:02 -0500 Subject: [Freeswitch-users] new version of vpx needed for those tracking master. Message-ID: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features. The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you are using our debian packages for dependencies, these packages are already available in the unstable repo. From vagarwal at vertical.com Thu Feb 18 21:49:45 2016 From: vagarwal at vertical.com (Varsha Agarwal) Date: Thu, 18 Feb 2016 18:49:45 +0000 Subject: [Freeswitch-users] uuid_simplify or uuid_deflect in mod_callCenter Message-ID: <04AB0A185A23864CA9255FD38901F65D023374A69A@SCEX1.vertical.com> Hi, Can I use uuid_simplify or uuid_deflect in CallCenter config to route agent calls? I want to send REFER with Replace for Agent transfers as our agents are registered to a different PBX and FreeSwitch is just a gateway. REFER will remove FreeSwitch from the call path. If yes, can I get an example, I tried using it in CallCenter config but I get error cannot locate uuid_simplify or cannot locate channel uuid_deflect. Thanks, Varsha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/8f13741e/attachment.html From mafra.fabio at gmail.com Thu Feb 18 22:02:12 2016 From: mafra.fabio at gmail.com (Fabio Mafra) Date: Thu, 18 Feb 2016 17:02:12 -0200 Subject: [Freeswitch-users] Persistent caller id on the download links ... Message-ID: I am a new user FreeSWITCH. I found a flaw in the system right after installation to make a call handover between 3 SIP extensions. Example: Extension 1 Extension 2 calls. 2 extension meets and press the "Hold" button. Extension 2 press the "transfer" button, and dials the number of the extension 3. Extension 2 and Extension 3 communicate. Extension 2 press the "transfer" button again to complete the transfer of extension 1 to extension 3. Pressing the "transfer" the extension 2 comes out link. Now Extension 1 and Extension 3 are communicating connection with the extension 2. But in Extension 1 and Extension Display 3 appear as if they were communicating with Extension 2 (no longer part of the link). After transfer to the Station 1 does not have any indication be communicating with the Station 3? After the transfer of the branch line 3 does not have any indication be communicating with Extension 1? How can I fix this problem? My FreeSWITCH is installed on a Debian machine 8-Jessie at: / usr / local / freeswitch. I use the internal settings of extensions 1000.xml to 1019.xml. Thank you for your attention right now! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/045c09a1/attachment.html From mario_fs at mgtech.com Thu Feb 18 22:56:20 2016 From: mario_fs at mgtech.com (Mario G) Date: Thu, 18 Feb 2016 11:56:20 -0800 Subject: [Freeswitch-users] new version of vpx needed for those tracking master. In-Reply-To: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> Message-ID: FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too? > On Feb 18, 2016, at 8:28 AM, Michael Jerris wrote: > > Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features. The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you are using our debian packages for dependencies, these packages are already available in the unstable repo. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Thu Feb 18 23:07:38 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 18 Feb 2016 15:07:38 -0500 Subject: [Freeswitch-users] new version of vpx needed for those tracking master. In-Reply-To: References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> Message-ID: I have not updated it yet.. will need to address that soon. > On Feb 18, 2016, at 2:56 PM, Mario G wrote: > > FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too? > > >> On Feb 18, 2016, at 8:28 AM, Michael Jerris wrote: >> >> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features. The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you are using our debian packages for dependencies, these packages are already available in the unstable repo. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 mario_fs at mgtech.com Fri Feb 19 01:27:46 2016 From: mario_fs at mgtech.com (Mario G) Date: Thu, 18 Feb 2016 14:27:46 -0800 Subject: [Freeswitch-users] new version of vpx needed for those tracking master. In-Reply-To: References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> Message-ID: <15C74368-4311-40B8-A30C-1244A9C4917B@mgtech.com> OK thank you, if you let me know I?ll test the 3 releases afterwards. Mario G > On Feb 18, 2016, at 12:07 PM, Michael Jerris wrote: > > I have not updated it yet.. will need to address that soon. > >> On Feb 18, 2016, at 2:56 PM, Mario G wrote: >> >> FS build os now broken on OS X, I noticed that home-brew did not update libvpx. Assume it needs to be updated too? >> >> >>> On Feb 18, 2016, at 8:28 AM, Michael Jerris wrote: >>> >>> Please note, we have updated code in master that will now require the newer version of libvpx (and anything such as libav that is linked against vpx will also need to be updated) in order to use video features. The tarball for this newer version is located at http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you are using our debian packages for dependencies, these packages are already available in the unstable repo. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 gascagonzalo at gmail.com Fri Feb 19 07:59:51 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Thu, 18 Feb 2016 20:59:51 -0800 Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator (self-signed certificates) Message-ID: I compiled FS 1.6 from source and using Verto Communicator with *Firefox* shows: "Waiting for server reconnection." Firefox 44.0.2 - MacOS system 10.10.4. - Windows 7 -*Works fine with Chrome* (I tested Chrome successfully to confirm there is no network issues). *-Same error in 2 different networks.* It works only after I enter in Browser manually: https://:8082 and accept the certificate I'm using self-signed certificates. Is this Browser related or Application related? Looks like is Browser related, need confirmation. *Console logs:* http://pastebin.com/XSJA9dZf *References:* https://bugzilla.mozilla.org/show_bug.cgi?id=594502 https://freeswitch.org/jira/browse/FS-8805 [image: Inline image 1] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/d1ddd19b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 134816 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/d1ddd19b/attachment-0001.png From gmaruzz at gmail.com Fri Feb 19 09:33:33 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 19 Feb 2016 07:33:33 +0100 Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator (self-signed certificates) In-Reply-To: References: Message-ID: As very clearly put in all documentation, self signed certificates are NOT supported (they may work if you really know how to do). Use real certificates. Google "letsencrypt" and use their utility to get real certificates completely for free. -giovanni Il 19/Feb/2016 06:01, "Gonzalo Gasca Meza" ha scritto: > I compiled FS 1.6 from source and using Verto Communicator with *Firefox* > shows: > "Waiting for server reconnection." > Firefox 44.0.2 > > - MacOS system 10.10.4. > - Windows 7 > > > > -*Works fine with Chrome* (I tested Chrome successfully to confirm there > is no network issues). > *-Same error in 2 different networks.* > > It works only after I enter in Browser manually: > https://:8082 and accept the certificate > > I'm using self-signed certificates. Is this Browser related or Application > related? > Looks like is Browser related, need confirmation. > > *Console logs:* > http://pastebin.com/XSJA9dZf > > > *References:* > https://bugzilla.mozilla.org/show_bug.cgi?id=594502 > https://freeswitch.org/jira/browse/FS-8805 > > > > [image: Inline image 1] > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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... 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Name: image.png Type: image/png Size: 134816 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/c369cf82/attachment-0001.png From gascagonzalo at gmail.com Fri Feb 19 09:43:30 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Thu, 18 Feb 2016 22:43:30 -0800 Subject: [Freeswitch-users] Firefox vs Chrome in Verto Communicator (self-signed certificates) In-Reply-To: References: Message-ID: Thanks Giovanni On Thu, Feb 18, 2016 at 10:33 PM, Giovanni Maruzzelli wrote: > As very clearly put in all documentation, self signed certificates are NOT > supported (they may work if you really know how to do). > > Use real certificates. > > Google "letsencrypt" and use their utility to get real certificates > completely for free. > > -giovanni > Il 19/Feb/2016 06:01, "Gonzalo Gasca Meza" ha > scritto: > >> I compiled FS 1.6 from source and using Verto Communicator with *Firefox* >> shows: >> "Waiting for server reconnection." >> Firefox 44.0.2 >> >> - MacOS system 10.10.4. >> - Windows 7 >> >> >> >> -*Works fine with Chrome* (I tested Chrome successfully to confirm there >> is no network issues). >> *-Same error in 2 different networks.* >> >> It works only after I enter in Browser manually: >> https://:8082 and accept the certificate >> >> I'm using self-signed certificates. Is this Browser related or >> Application related? >> Looks like is Browser related, need confirmation. >> >> *Console logs:* >> http://pastebin.com/XSJA9dZf >> >> >> *References:* >> https://bugzilla.mozilla.org/show_bug.cgi?id=594502 >> https://freeswitch.org/jira/browse/FS-8805 >> >> >> >> [image: Inline image 1] >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160218/301d865a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 134816 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160218/301d865a/attachment-0001.png From aqsyounas at gmail.com Fri Feb 19 13:40:09 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 19 Feb 2016 15:40:09 +0500 Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! Message-ID: Hi. I am able to compile mod_avmd without any error. But when i try to load it i see this error in console. freeswitch at internal> load mod_avmd +OK Reloading XML -ERR [module load file routine returned an error] 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_avmd.so **Module load routine returned an error** These are the lines I see in mod_avmd.c but unable to understand. if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); return SWITCH_STATUS_TERM; } Any pointer is much appreciated. Best Regards. This email has been sent from a virus-free computer protected by Avast. www.avast.com <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/b3c3bfd8/attachment.html From vma at 440hz.fr Fri Feb 19 14:33:10 2016 From: vma at 440hz.fr (Vallimamod Abdullah) Date: Fri, 19 Feb 2016 12:33:10 +0100 Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! In-Reply-To: References: Message-ID: <22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr> Hi, Looks like you are trying to load a module that is already loaded. Try a reload. Also, sometimes the event subclass doesn?t unregister properly when you do a reload so you need to restart freeswitch when you get this error. Best Regards, Vallimamod . > On 19 Feb 2016, at 11:40, Aqs Younas wrote: > > Hi. > > I am able to compile mod_avmd without any error. But when i try to load it i see this error in console. > > freeswitch at internal> load mod_avmd > +OK Reloading XML > -ERR [module load file routine returned an error] > > 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded > 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! > 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_avmd.so > **Module load routine returned an error** > > > These are the lines I see in mod_avmd.c but unable to understand. > > > if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); > return SWITCH_STATUS_TERM; > } > > > Any pointer is much appreciated. > > Best Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/01bb2f39/attachment.html From aqsyounas at gmail.com Fri Feb 19 14:56:55 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 19 Feb 2016 16:56:55 +0500 Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! In-Reply-To: <22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr> References: <22E5B8B0-1C62-4BD1-9B94-32ECF3D5E0A5@440hz.fr> Message-ID: Perfect. Thank you. ! This email has been sent from a virus-free computer protected by Avast. www.avast.com <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On 19 February 2016 at 16:33, Vallimamod Abdullah wrote: > Hi, > > Looks like you are trying to load a module that is already loaded. Try a > reload. Also, sometimes the event subclass doesn?t unregister properly when > you do a reload so you need to restart freeswitch when you get this error. > > Best Regards, > Vallimamod > . > > > On 19 Feb 2016, at 11:40, Aqs Younas wrote: > > Hi. > > I am able to compile mod_avmd without any error. But when i try to load it > i see this error in console. > > freeswitch at internal> load mod_avmd > +OK Reloading XML > -ERR [module load file routine returned an error] > > 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded > 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass > avmd::beep! > 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error > Loading module /usr/local/freeswitch/mod/mod_avmd.so > **Module load routine returned an error** > > > These are the lines I see in mod_avmd.c but unable to understand. > > > if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, > "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); > return SWITCH_STATUS_TERM; > } > > > Any pointer is much appreciated. > > 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/20160219/64f71610/attachment.html From brian at freeswitch.org Fri Feb 19 15:44:19 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Feb 2016 06:44:19 -0600 Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! In-Reply-To: References: Message-ID: [image: Inline image 1] On Fri, Feb 19, 2016 at 4:40 AM, Aqs Younas wrote: > Hi. > > I am able to compile mod_avmd without any error. But when i try to load it > i see this error in console. > > freeswitch at internal> load mod_avmd > +OK Reloading XML > -ERR [module load file routine returned an error] > > 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded > 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass > avmd::beep! > 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error > Loading module /usr/local/freeswitch/mod/mod_avmd.so > **Module load routine returned an error** > > > These are the lines I see in mod_avmd.c but unable to understand. > > > if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, > "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); > return SWITCH_STATUS_TERM; > } > > > Any pointer is much appreciated. > > Best Regards. > > This email has been sent from a virus-free computer protected by Avast. > www.avast.com > <#1121851626_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 > -- *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/20160219/ae81c6ec/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: dialer.png Type: image/png Size: 122976 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/ae81c6ec/attachment-0001.png From aqsyounas at gmail.com Fri Feb 19 15:59:53 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 19 Feb 2016 17:59:53 +0500 Subject: [Freeswitch-users] [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! In-Reply-To: References: Message-ID: I was using NewFies Dialer :) On 19 February 2016 at 17:44, Brian West wrote: > [image: Inline image 1] > > > On Fri, Feb 19, 2016 at 4:40 AM, Aqs Younas wrote: > >> Hi. >> >> I am able to compile mod_avmd without any error. But when i try to load >> it i see this error in console. >> >> freeswitch at internal> load mod_avmd >> +OK Reloading XML >> -ERR [module load file routine returned an error] >> >> 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded >> 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register >> subclass avmd::beep! >> 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error >> Loading module /usr/local/freeswitch/mod/mod_avmd.so >> **Module load routine returned an error** >> >> >> These are the lines I see in mod_avmd.c but unable to understand. >> >> >> if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != >> SWITCH_STATUS_SUCCESS) { >> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, >> "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); >> return SWITCH_STATUS_TERM; >> } >> >> >> Any pointer is much appreciated. >> >> Best Regards. >> >> This email has been sent from a virus-free computer protected by Avast. >> www.avast.com >> <#-55043839_1121851626_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 >> > > > > -- > > *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/20160219/837b46f6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: dialer.png Type: image/png Size: 122976 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/837b46f6/attachment-0001.png From areski at gmail.com Fri Feb 19 17:39:16 2016 From: areski at gmail.com (Areski) Date: Fri, 19 Feb 2016 15:39:16 +0100 Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana, InfluxDB & Telegraf Message-ID: Hi, If you are interested how to build this Live monitoring system for FreeSWITCH using Grafana & InfluxDB, we have written a post describing how: http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/ I would love to hear how you do your own monitoring or how we could improve this. -- Kind regards, /Areski ---- Arezqui Belaid, Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/8f9dbe9f/attachment.html From gmaruzz at gmail.com Fri Feb 19 17:53:35 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 19 Feb 2016 15:53:35 +0100 Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana, InfluxDB & Telegraf In-Reply-To: References: Message-ID: Yay Areski!!!!! Looking forward to read that beauty ! -giovanni Il 19/Feb/2016 15:40, "Areski" ha scritto: > Hi, > > If you are interested how to build this Live monitoring system for > FreeSWITCH using Grafana & InfluxDB, we have written a post describing how: > > http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/ > > I would love to hear how you do your own monitoring or how we could > improve this. > > -- > Kind regards, > /Areski > > ---- > Arezqui Belaid, > Founder at Star2Billing (www.star2billing.com) > > Tel: +34650784355 > Twitter: http://twitter.com/areskib > LinkedIn: http://www.linkedin.com/in/areski > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160219/e549cf38/attachment.html From gmaruzz at gmail.com Fri Feb 19 18:22:53 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 19 Feb 2016 16:22:53 +0100 Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana, InfluxDB & Telegraf In-Reply-To: References: Message-ID: Very nice post and solution! Looking forward to install and test it. We use Homer (Voipcapture) + check_mk -giovanni On Fri, Feb 19, 2016 at 3:39 PM, Areski wrote: > Hi, > > If you are interested how to build this Live monitoring system for > FreeSWITCH using Grafana & InfluxDB, we have written a post describing how: > > http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/ > > I would love to hear how you do your own monitoring or how we could > improve this. > > -- > Kind regards, > /Areski > > ---- > Arezqui Belaid, > Founder at Star2Billing (www.star2billing.com) > > Tel: +34650784355 > Twitter: http://twitter.com/areskib > LinkedIn: http://www.linkedin.com/in/areski > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160219/55e95c5c/attachment.html From areski at gmail.com Fri Feb 19 18:57:37 2016 From: areski at gmail.com (Areski) Date: Fri, 19 Feb 2016 16:57:37 +0100 Subject: [Freeswitch-users] Live Monitoring FreeSWITCH with Grafana, InfluxDB & Telegraf In-Reply-To: References: Message-ID: Thanks for the feedback Giovanni, I will definitely look at Homer for quality monitoring, didnt had time to play with it yet :) What was really important for us now it's to provide a multitenant UI where users could see their own metrics but also measure vital application metrics like redis, postgresql, etc... On Fri, Feb 19, 2016 at 4:22 PM, Giovanni Maruzzelli wrote: > Very nice post and solution! Looking forward to install and test it. > > We use Homer (Voipcapture) + check_mk > > -giovanni > > On Fri, Feb 19, 2016 at 3:39 PM, Areski wrote: > >> Hi, >> >> If you are interested how to build this Live monitoring system for >> FreeSWITCH using Grafana & InfluxDB, we have written a post describing how: >> >> http://www.newfies-dialer.org/live-monitoring-freeswitch-newfies-dialer-with-grafana-influxdb-telegraf/ >> >> I would love to hear how you do your own monitoring or how we could >> improve this. >> >> -- >> Kind regards, >> /Areski >> >> ---- >> Arezqui Belaid, >> Founder at Star2Billing (www.star2billing.com) >> >> Tel: +34650784355 >> Twitter: http://twitter.com/areskib >> LinkedIn: http://www.linkedin.com/in/areski >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- Kind regards, /Areski ---- Arezqui Belaid, Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/35319bc1/attachment-0001.html From martin.hoole at emailn.de Fri Feb 19 19:31:46 2016 From: martin.hoole at emailn.de (martin.hoole at emailn.de) Date: Fri, 19 Feb 2016 17:31:46 +0100 Subject: [Freeswitch-users] Selective Forwarding Unit (SFU) Message-ID: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de> Are there plans to implement support for Selective Forwarding Unit (SFU) for video or audio rather than mixing video together? https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-01 From anthony.minessale at gmail.com Fri Feb 19 19:40:28 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Feb 2016 10:40:28 -0600 Subject: [Freeswitch-users] Selective Forwarding Unit (SFU) In-Reply-To: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de> References: <73dcf231ea663b1f127af8b41efcb108@mail.emailn.de> Message-ID: Not without a massive demand for it from the user base and appropriate funding to spend the resources on it. Otherwise, you can just use Jitsi Video Bridge. On Fri, Feb 19, 2016 at 10:31 AM, wrote: > Are there plans to implement support for Selective Forwarding Unit (SFU) > for video or audio rather than mixing video together? > https://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-01 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160219/2328d90d/attachment.html From deepikay at iiitd.ac.in Fri Feb 19 20:09:30 2016 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 19 Feb 2016 22:39:30 +0530 Subject: [Freeswitch-users] Library linking problem in ESL Message-ID: Hi, I am using java ESL, and getting the following error : Exception in thread "main" java.lang.UnsatisfiedLinkError:org.freeswitch.swig.freeswitchJNI.new_JavaSession_SWIG_0()J at org.freeswitch.swig.freeswitchJNI.new_JavaSession__SWIG_0(Native Method) at org.freeswitch.swig.JavaSession.(JavaSession.java:37) at MyESLTest.main(MyESLTest.java:465) and I also activates another similar esl program from the dialplan that runs fine. would be thankful if any clue regarding resolving it can be suggested. Regards, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160219/ea6835f4/attachment.html From tomasz.o.ostrowski at gmail.com Fri Feb 19 22:24:40 2016 From: tomasz.o.ostrowski at gmail.com (Tomasz Ostrowski) Date: Fri, 19 Feb 2016 20:24:40 +0100 Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE Message-ID: Hello, while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored problem when FreeSWITCH replies with SDP containing: m=image 0 udptl t38 m=image 0 udptl t38 when receiving re-INVITE with disabled audio media and enabled image media (switching from audio to image). As far as I know this is correct way to change media, from RFC 3264: 8.1 Adding a Media Stream New media streams are created by new additional media descriptions below the existing ones, or by reusing the "slot" used by an old media stream which had been disabled by setting its port to zero. Reusing its slot means that the new media description replaces the old one, but retains its positioning relative to other media descriptions in the SDP. New media descriptions MUST appear below any existing media sections. The rules for formatting these media descriptions are identical to those described in Section 5. When the answerer receives an SDP with more media descriptions than the previous SDP from the offerer, or it receives an SDP with a media stream in a slot where the port was previously zero, the answerer knows that new media streams are being added. These can be rejected or accepted by placing an appropriately structured media description in the answer. The procedures for constructing the new media description in the answer are described in Section 6. This problem is mentioned in: https://freeswitch.org/jira/browse/FS-7037 https://freeswitch.org/jira/browse/FS-6212 With reversed transmission direction (when FreeSWITCH receives FAX and re-INVITES) re-INVITE contains only image media, but this is easier to accept (https://freeswitch.org/jira/browse/FS-6954 - correcting it caused interoperability problems between FS versions), while not accepting correct SDP by FreeSWITCH is really painful as it requires implementing RFC non-compliant negotiation and probably adding special switch in configuration to be interoperable. Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further references in jira)? Could you give me any suggestions where to look in FreeSWITCH source code (10k LOC sofia.c seems pretty complex)? -- TMSZ From Jerry.Kendall at FiberConX.com Fri Feb 19 22:28:19 2016 From: Jerry.Kendall at FiberConX.com (Jerry Kendall) Date: Fri, 19 Feb 2016 14:28:19 -0500 Subject: [Freeswitch-users] UNALLOCATED_NUMBER handling Message-ID: <56C76CD3.3000008@FiberConX.com> Hello all.... I'm trying to add code to the dialplan to handle the UNALLOCATED_NUMBER error. I'm in Canada, when I call a US based 800 number I should, and now am getting UNALLOCATED_NUMBER error from the Canadian based ITSP. However, I want to handle it in the DialPlan. I want to tell the users, via audio file playback, that the number is not assigned/allocated. I have looked in lots of places and found references to transfer_on_fail and a few other ideas and have tried many but, I just can't seem to make this work. Not sure if this is the best way to handle this. my 'default' dialplan has this------------- and I have the following in for UNALLOCATED_NUMBER extension---- Jerry From mike at jerris.com Fri Feb 19 23:00:05 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 Feb 2016 15:00:05 -0500 Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE In-Reply-To: References: Message-ID: <43B10980-E729-4087-ABA5-668C6F610A69@jerris.com> I think we fixed this in 1.6 already. Give it a try to confirm but i recall this one getting fixed. > On Feb 19, 2016, at 2:24 PM, Tomasz Ostrowski wrote: > > Hello, > while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored > problem when FreeSWITCH replies with SDP containing: > > m=image 0 udptl t38 > m=image 0 udptl t38 > > when receiving re-INVITE with disabled audio media and enabled image media > (switching from audio to image). As far as I know this is correct way to > change media, from RFC 3264: > > 8.1 Adding a Media Stream > > New media streams are created by new additional media descriptions > below the existing ones, or by reusing the "slot" used by an old > media stream which had been disabled by setting its port to zero. > > Reusing its slot means that the new media description replaces the > old one, but retains its positioning relative to other media > descriptions in the SDP. New media descriptions MUST appear below > any existing media sections. The rules for formatting these media > descriptions are identical to those described in Section 5. > > When the answerer receives an SDP with more media descriptions than > the previous SDP from the offerer, or it receives an SDP with a media > stream in a slot where the port was previously zero, the answerer > knows that new media streams are being added. These can be rejected > or accepted by placing an appropriately structured media description > in the answer. The procedures for constructing the new media > description in the answer are described in Section 6. > > This problem is mentioned in: > https://freeswitch.org/jira/browse/FS-7037 > https://freeswitch.org/jira/browse/FS-6212 > > With reversed transmission direction (when FreeSWITCH receives FAX and > re-INVITES) re-INVITE contains only image media, but this is easier to > accept (https://freeswitch.org/jira/browse/FS-6954 - correcting it caused > interoperability problems between FS versions), while not accepting > correct SDP by FreeSWITCH is really painful as it requires implementing > RFC non-compliant negotiation and probably adding special switch in > configuration to be interoperable. > > Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further > references in jira)? > Could you give me any suggestions where to look in FreeSWITCH source code > (10k LOC sofia.c seems pretty complex)? > > -- > TMSZ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Sat Feb 20 06:19:54 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 19 Feb 2016 21:19:54 -0600 Subject: [Freeswitch-users] one way audio - another report related to auto-jitter-buffer-msec possible behavior change with 1.6 Message-ID: <56C7DB5A.6030200@mst.edu> Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams) with 1.6. AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to FS, but FS wasn't passing any of it on to the provider. Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled, and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment. I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the grandstream (testing from home). -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From anthony.minessale at gmail.com Sat Feb 20 08:13:09 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Feb 2016 23:13:09 -0600 Subject: [Freeswitch-users] one way audio - another report related to auto-jitter-buffer-msec possible behavior change with 1.6 In-Reply-To: <56C7DB5A.6030200@mst.edu> References: <56C7DB5A.6030200@mst.edu> Message-ID: Proported bugs and details therein belong on Jira. On Friday, February 19, 2016, Nathan Neulinger wrote: > Just wanted to add an additional data point on the one way audio issue (as > it turns out - also with grandstreams) with 1.6. > > AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider > > One my most recent test/capture/etc, the behavior I saw was that it looked > to me like audio stopped being sent from > freeswitch to outside/external leg once early media finished. Note that > the grandstream was continuing to send RTP to > FS, but FS wasn't passing any of it on to the provider. > > Once I saw the thread from the other user having this problem, tried > turning it off on my profiles that had it enabled, > and the problem appears to have gone away. I can provide > logs/debug/captures/etc. if requested from a test environment. > > I did not have any reports of this issue except related to grandstreams > (ht-701) - in my case, almost all related to > analog faxes (bulk of our ATA usage). Also have not been able to verify > the fix yet on a call originating from the > grandstream (testing from home). > > -- 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 > -- 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/20160219/633d86b8/attachment.html From royj at yandex.ru Sat Feb 20 09:26:42 2016 From: royj at yandex.ru (royj at yandex.ru) Date: Sat, 20 Feb 2016 09:26:42 +0300 Subject: [Freeswitch-users] UNALLOCATED_NUMBER handling In-Reply-To: <56C76CD3.3000008@FiberConX.com> References: <56C76CD3.3000008@FiberConX.com> Message-ID: <1442451455949602@web18o.yandex.ru> I think it should work without that 'continue="true"' in 'extension name="public_did_11234567890"' and without 'hangup_after_bridge=true' 19.02.2016, 22:38, "Jerry Kendall" : > Hello all.... > > I'm trying to add code to the dialplan to handle the UNALLOCATED_NUMBER error. > > I'm in Canada, when I call a US based 800 number I should, and now am getting UNALLOCATED_NUMBER > error from the Canadian based ITSP. > However, I want to handle it in the DialPlan. I want to tell the users, via audio file playback, > that the number is not assigned/allocated. > > I have looked in lots of places and found references to transfer_on_fail and a few other ideas and > have tried many but, I just can't seem to make this work. > > Not sure if this is the best way to handle this. > > my 'default' dialplan has this------------- > > ????????? > ????????? > ????????????????? > ????????????????? > > ????????????????????? > ????????????????????? > ????????????????????? > > ????????????????????? > > ????????????????? > > ????????? > > > and I have the following in for UNALLOCATED_NUMBER extension---- > > ????????? > ????????????????? continue="false" break="on-true"> > ????????????????????????? > ????????????????????????? data="$${conf_dir}/sounds/unallocated_number.wav"/> > ????????????????????????? > ????????????????? > ????????? > > > Jerry > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 s.safarov at gmail.com Sat Feb 20 12:44:08 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 20 Feb 2016 12:44:08 +0300 Subject: [Freeswitch-users] new version of vpx needed for those tracking master. In-Reply-To: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> References: <6F83920F-35F5-4B15-B9CC-AC5C4648C7DA@jerris.com> Message-ID: Creaed PR4 to allow rpm package creation On Thu, Feb 18, 2016 at 7:28 PM, Michael Jerris wrote: > Please note, we have updated code in master that will now require the > newer version of libvpx (and anything such as libav that is linked against > vpx will also need to be updated) in order to use video features. The > tarball for this newer version is located at > http://files.freeswitch.org/downloads/libs/libvpx2-1.5.0.tar.gz . If you > are using our debian packages for dependencies, these packages are already > available in the unstable repo. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160220/b2419a05/attachment.html From bilaln018 at gmail.com Sat Feb 20 16:08:29 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 20 Feb 2016 18:08:29 +0500 Subject: [Freeswitch-users] [mod_skyopen][Debian Jessie][Error while loading] Message-ID: Hi Users, I am getting error while loading mod_skyopen on debian jessie 8 64bit, i am using freeswitch Version 1.7.0 git 09b4156 2016-02-04 17:08:30Z 64bit Logs: https://pastebin.freeswitch.org/24569 I could not find any official link for installation on Debian 8. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160220/5a9df581/attachment.html From bilaln018 at gmail.com Sat Feb 20 16:09:31 2016 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 20 Feb 2016 18:09:31 +0500 Subject: [Freeswitch-users] Skypopen Debian Jessie install issues In-Reply-To: References: Message-ID: Did you got any fix? I am facing the same. Regards Abbasi On Wed, Dec 16, 2015 at 12:50 AM, Sergey Safarov wrote: > Please read > http://lists.freeswitch.org/pipermail/freeswitch-users/2015-November/117273.html > > On Tue, Dec 15, 2015 at 3:08 PM, Jason Bedward wrote: > >> Hi, >> >> I have tried to install this on Debian Jessie but had some issues with >> dependencies which are not matching what the install guide says. When I got >> to the CLI and load the module I get the error below. I >> have libjpeg62-turbo & libjpeg62-turbo-dev installed which I think it the >> current version for Debian. Please can someone help guide me with this >> version of Debian. >> >> Thanks >> >> >> +OK log level 9 [9] >> +OK console log level set to DEBUG >> >> freeswitch at 192.168.2.233@internal> load mod_skypopen >> +OK Reloading XML >> -ERR [module load file routine returned an error] >> >> 2015-12-15 12:01:55.996236 [INFO] mod_enum.c:880 ENUM Reloaded >> 2015-12-15 12:01:55.996236 [INFO] switch_time.c:1415 Timezone reloaded >> 1781 definitions >> 2015-12-15 12:01:55.996236 [CRIT] switch_loadable_module.c:1447 Error >> Loading module /usr/local/freeswitch/mod/mod_skypopen.so >> **/usr/local/freeswitch/mod/mod_skypopen.so: undefined symbol: >> jpeg_resync_to_restart** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160220/e6bd56ae/attachment-0001.html From nneul at mst.edu Sat Feb 20 16:48:31 2016 From: nneul at mst.edu (Nathan Neulinger) Date: Sat, 20 Feb 2016 07:48:31 -0600 Subject: [Freeswitch-users] one way audio - another report related to auto-jitter-buffer-msec possible behavior change with 1.6 In-Reply-To: References: <56C7DB5A.6030200@mst.edu> Message-ID: <56C86EAF.9060402@mst.edu> https://freeswitch.org/jira/browse/FS-8850 On 02/19/2016 11:13 PM, Anthony Minessale wrote: > Proported bugs and details therein belong on Jira. > > On Friday, February 19, 2016, Nathan Neulinger > wrote: > > Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams) > with 1.6. > > AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider > > One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from > freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to > FS, but FS wasn't passing any of it on to the provider. > > Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled, > and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment. > > I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to > analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the > grandstream (testing from home). > > -- 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 > > > > -- > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From s.safarov at gmail.com Sat Feb 20 17:24:01 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 20 Feb 2016 17:24:01 +0300 Subject: [Freeswitch-users] recording problems with mod_shout debian/master In-Reply-To: References: Message-ID: Detailed description of error is given on FS-8851 Sergey On Mon, Feb 1, 2016 at 4:47 PM, 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/20160220/048fa280/attachment.html From anthony.minessale at gmail.com Sat Feb 20 20:08:49 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Feb 2016 11:08:49 -0600 Subject: [Freeswitch-users] recording problems with mod_shout debian/master In-Reply-To: References: Message-ID: Does kazoo have bug reports? They are probably starting the call recording too soon. On Saturday, February 20, 2016, Sergey Safarov wrote: > Detailed description of error is given on FS-8851 > > Sergey > > On Mon, Feb 1, 2016 at 4:47 PM, 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 >> 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 >> * >> 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 >> > > -- 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/20160220/7ac2ad08/attachment.html From d at d-man.org Sat Feb 20 20:23:10 2016 From: d at d-man.org (Darren) Date: Sat, 20 Feb 2016 17:23:10 +0000 Subject: [Freeswitch-users] recording problems with mod_shout debian/master In-Reply-To: References: Message-ID: <77E4928D-EC8E-40D9-97FD-9562B6AA0B26@d-man.org> Can you please file this at tickets.2600hz.com and we'll take a look? This seems odd to me. From: > on behalf of Anthony Minessale > Reply-To: FreeSWITCH Users Help > Date: Sunday, February 21, 2016 at 6:08 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] recording problems with mod_shout debian/master Does kazoo have bug reports? They are probably starting the call recording too soon. On Saturday, February 20, 2016, Sergey Safarov > wrote: Detailed description of error is given on FS-8851 Sergey On Mon, Feb 1, 2016 at 4:47 PM, 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 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 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 -- 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/20160220/6109eaed/attachment-0001.html From gregor at infomedia.si Sun Feb 21 02:00:25 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 21 Feb 2016 00:00:25 +0100 Subject: [Freeswitch-users] Verto Message-ID: Hi! Haven't try Verto yet, but just wondering what happens if I login in multiple browser window with same username? What will happen on outgoing or incoming call? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160221/d8ff49d2/attachment.html From krice at freeswitch.org Sun Feb 21 04:58:40 2016 From: krice at freeswitch.org (Ken Rice) Date: Sat, 20 Feb 2016 19:58:40 -0600 Subject: [Freeswitch-users] Verto In-Reply-To: References: Message-ID: <23ca01d16c4b$63206730$29613590$@freeswitch.org> This depends on several factors? but it is workable in many configs this way?. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger Sent: Saturday, February 20, 2016 5:00 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Verto Hi! Haven't try Verto yet, but just wondering what happens if I login in multiple browser window with same username? What will happen on outgoing or incoming call? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160220/96964e5f/attachment.html From s.safarov at gmail.com Sun Feb 21 06:26:57 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 21 Feb 2016 03:26:57 +0000 Subject: [Freeswitch-users] Free learning WebRTC book Message-ID: If you interested, please download WebRTC book https://www.packtpub.com/packt/offers/free-learning -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160221/b53e6c80/attachment.html From gregor at infomedia.si Sun Feb 21 09:17:46 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 21 Feb 2016 07:17:46 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <23ca01d16c4b$63206730$29613590$@freeswitch.org> References: <23ca01d16c4b$63206730$29613590$@freeswitch.org> Message-ID: Guess it is same as login with softphone with same credentials. It would be best to try and see :-) 2016-02-21 2:58 GMT+01:00 Ken Rice : > This depends on several factors? but it is workable in many configs this > way?. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor > Nanger > *Sent:* Saturday, February 20, 2016 5:00 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Verto > > > > Hi! > > > > Haven't try Verto yet, but just wondering what happens if I login in > multiple browser window with same username? What will happen on outgoing or > incoming call? > > > > Best regards, Gregor > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160221/03820e15/attachment.html From krice at freeswitch.org Sun Feb 21 09:31:25 2016 From: krice at freeswitch.org (Ken Rice) Date: Sun, 21 Feb 2016 00:31:25 -0600 Subject: [Freeswitch-users] Verto In-Reply-To: References: <23ca01d16c4b$63206730$29613590$@freeswitch.org> Message-ID: <798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org> My cryptic reply wasnt meant to be cryptic, its just it really depends on his configs and how he sets it up Same browser different tabs works differently if depending on old school standard move vs private browsing mode and may depend on how the things are even configured... His question doesnt have enough specifics for a direct answer Sent from my iPhone > On Feb 21, 2016, at 12:17 AM, Gregor Nanger wrote: > > Guess it is same as login with softphone with same credentials. It would be best to try and see :-) > > 2016-02-21 2:58 GMT+01:00 Ken Rice : >> This depends on several factors? but it is workable in many configs this way?. >> >> >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger >> Sent: Saturday, February 20, 2016 5:00 PM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] Verto >> >> >> >> Hi! >> >> >> >> Haven't try Verto yet, but just wondering what happens if I login in multiple browser window with same username? What will happen on outgoing or incoming call? >> >> >> >> Best regards, Gregor >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160221/b7c07d3d/attachment-0001.html From alhakeem at gmail.com Sun Feb 21 15:34:09 2016 From: alhakeem at gmail.com (Abdul Hakeem) Date: Sun, 21 Feb 2016 12:34:09 +0000 Subject: [Freeswitch-users] Verto & Sofia Registrations Message-ID: Hello, My understanding is that Sofia can only do one registration at a time, does Veryo exhibit this limit ? Is there any way to process multiple registrations in the background or to configure FS to use the database of another REGISTRAR ?. Much obliged, AH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160221/3b4c1061/attachment.html From adahary at gmail.com Sun Feb 21 15:58:42 2016 From: adahary at gmail.com (Assaf Dahary) Date: Sun, 21 Feb 2016 14:58:42 +0200 Subject: [Freeswitch-users] Freeswitch extended with ZRTP Message-ID: <030f01d16ca7$98240250$c86c06f0$@gmail.com> Hi, I would like to setup FS1 as a 'ZRTP gateway' to a none-zrtp sip client. Here is my net setup: Client1 (without ZRTP)-> LAN -> FS1 (Gateway: register to FS2) -> NAT -> Internet-> Public IP (not NAT) FS2 -> CSipSimple (ZRTP enabled). Client1: ? Ex#1000, Registered over LAN to FS1. FS1: ? Being NAT with dynamic IP address ? Registered as a Gateway to FS2 (in Internal profile). So FS1 is extended on FS2. ? Setup as 'Proxy-Media = false' && zrtp_enrollment=true (trusted MITM). FS2: ? Connected with static public IP address (not behind NAT) ? Setup as 'Proxy-Media = true' && inbound-late-negotiation=true. CSipSimple: ? Behind NAT (remote WiFi/3G) and is registered on FS2 (Internal profile). FS2 is successfully serving multiple CSipSimple ZRTP clients with end-to-end ZRTP secure calls. The problem: When calling from Client1/FS1 to FS2/CSipSimple, then FS1 shows ' WARNING! Incoming ZRTP CRC validation fails' and FS2 shows ' ZRTP not negotiated on both sides; disabling ZRTP passthr '. I have tested FS1 locally with CSipSimple and it manages to connect with ZRTP/MITM so it is capable of ZRTP. I follow up what have been recommended on the forum to extend FS1 and to verify matching codecs (I forced PCMU only on all devices) and to check RTP/UDP ports flow (SIP trace). I would appreciate any help on how to setup end-to-end ZRTP calls between FS1 and remote CsipSimple (FS2). Regards Assaf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160221/5c5e9711/attachment.html From krice at freeswitch.org Sun Feb 21 19:07:51 2016 From: krice at freeswitch.org (Ken Rice) Date: Sun, 21 Feb 2016 10:07:51 -0600 Subject: [Freeswitch-users] Verto & Sofia Registrations In-Reply-To: References: Message-ID: Sofia can do multi-reg its a setting on the profile Sent from my iPhone > On Feb 21, 2016, at 6:34 AM, Abdul Hakeem wrote: > > Hello, > My understanding is that Sofia can only do one registration at a time, does Veryo exhibit this limit ? > > Is there any way to process multiple registrations in the background or to configure FS to use the database of another REGISTRAR ?. > > Much obliged, > > AH > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160221/014e6806/attachment.html From gregor at infomedia.si Sun Feb 21 23:19:20 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 21 Feb 2016 21:19:20 +0100 Subject: [Freeswitch-users] Verto In-Reply-To: <798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org> References: <23ca01d16c4b$63206730$29613590$@freeswitch.org> <798CC583-BA4F-4CE1-BA83-264146ACC965@freeswitch.org> Message-ID: Thank you Ken, it is ok... I just have to try to have more specific question. 2016-02-21 7:31 GMT+01:00 Ken Rice : > My cryptic reply wasnt meant to be cryptic, its just it really depends on > his configs and how he sets it up > > Same browser different tabs works differently if depending on old school > standard move vs private browsing mode and may depend on how the things are > even configured... His question doesnt have enough specifics for a direct > answer > > Sent from my iPhone > > On Feb 21, 2016, at 12:17 AM, Gregor Nanger wrote: > > Guess it is same as login with softphone with same credentials. It would > be best to try and see :-) > > 2016-02-21 2:58 GMT+01:00 Ken Rice : > >> This depends on several factors? but it is workable in many configs this >> way?. >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor >> Nanger >> *Sent:* Saturday, February 20, 2016 5:00 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Verto >> >> >> >> Hi! >> >> >> >> Haven't try Verto yet, but just wondering what happens if I login in >> multiple browser window with same username? What will happen on outgoing or >> incoming call? >> >> >> >> Best regards, Gregor >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > -- 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/20160221/ba880af7/attachment-0001.html From Harald.Petrovitsch at sermotec.at Sun Feb 21 23:57:00 2016 From: Harald.Petrovitsch at sermotec.at (Harald Petrovitsch) Date: Sun, 21 Feb 2016 20:57:00 +0000 Subject: [Freeswitch-users] Freeswitch with freetdm on Windows Message-ID: Hi, is anybody out there using Freeswitch with Sangoma ISDN E1 Hardware on Windows in a productive environment ? Any stability Issues ? Which Freeswitch and Wanpipe version are you using ? Regards Harald -- harald.petrovitsch at sermotec.at From nandy1925 at gmail.com Mon Feb 22 01:24:05 2016 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 22 Feb 2016 06:24:05 +0800 Subject: [Freeswitch-users] Free learning WebRTC book In-Reply-To: References: Message-ID: Thanks Sergey :-) On Sun, Feb 21, 2016 at 11:26 AM, Sergey Safarov wrote: > If you interested, please download WebRTC book > https://www.packtpub.com/packt/offers/free-learning > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160222/e99d8613/attachment.html From rutu.patel at inextrix.com Mon Feb 22 09:19:29 2016 From: rutu.patel at inextrix.com (Rutu Patel) Date: Mon, 22 Feb 2016 11:49:29 +0530 Subject: [Freeswitch-users] Call dropping after 32 seconds Message-ID: Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160222/5450c6c6/attachment-0001.html -------------- next part -------------- U x.x.x.174:5060 -> x.x.x.166:5060 INVITE sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313. From: ;tag=1037646889. To: . Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 70 INVITE. Contact: . Max-Forwards: 30. User-Agent: Grandstream GXP1400 1.0.6.11. Privacy: none. P-Preferred-Identity: . Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 285. X-AUTH-IP: x.x.x.47. . v=0. o=xxxx9 8001 8000 IN IP4 x.x.x.47. s=SIP Call. c=IN IP4 x.x.x.47. t=0 0. m=audio 5008 RTP/AVP 0 8 18 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313. Record-Route: . From: ;tag=1037646889. To: . Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 70 INVITE. User-Agent: M01. Content-Length: 0. . U x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1606798992;rport=2313. From: ;tag=1037646889. To: ;tag=reUey9taa7r0N. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 70 INVITE. User-Agent: M01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Proxy-Authenticate: Digest realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", algorithm=MD5, qop="auth". Content-Length: 0. . U x.x.x.174:5060 -> x.x.x.166:5060 ACK sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK1bf1.f4e1b863.0. From: ;tag=1037646889. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. To: ;tag=reUey9taa7r0N. CSeq: 70 ACK. Max-Forwards: 70. Content-Length: 0. . U x.x.x.174:5060 -> x.x.x.166:5060 INVITE sip:xxxxxxx29858 at x.x.x.166:5060 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313. From: ;tag=1037646889. To: . Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 INVITE. Contact: . Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b. Max-Forwards: 30. User-Agent: Grandstream GXP1400 1.0.6.11. Privacy: none. P-Preferred-Identity: . Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 285. X-AUTH-IP: x.x.x.47. . v=0. o=xxxx9 8001 8000 IN IP4 x.x.x.47. s=SIP Call. c=IN IP4 x.x.x.47. t=0 0. m=audio 5008 RTP/AVP 0 8 18 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313. Record-Route: . From: ;tag=1037646889. To: . Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 INVITE. User-Agent: M01. Content-Length: 0. . U x.x.x.166:5060 -> x.x.x.174:5060 INVITE sip:xxxxxxx29858 at x.x.x.3 SIP/2.0. Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK789FQF04S4yte. Max-Forwards: 29. From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: . Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 324. X-AUTH-IP: x.x.x.47. Remote-Party-ID: "xxxxxxx00181" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1455243062 1455243063 IN IP4 x.x.x.166. s=FreeSWITCH. c=IN IP4 x.x.x.166. t=0 0. m=audio 27972 RTP/AVP 8 0 9 3 18 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP x.x.x.166;received=x.x.x.166;rport=5060;branch=z9hG4bK789FQF04S4yte. From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: . Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Content-Length: 0. Warning: 392 x.x.x.174:5060 "Noisy feedback tells: pid=36603 req_src_ip=x.x.x.166 req_src_port=5060 in_uri=sip:xxxxxxx29858 at x.x.x.3 out_uri=sip:xxxxxxx29858 at x.x.x.3 via_cnt==1". . U x.x.x.3:5060 -> x.x.x.166:5060 INVITE sip:xxxxxxx29858 at x.x.x.166 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0. Max-Forwards: 25. From: "xxxxxxx00181" ;tag=9pp9134ya08pe. To: . Call-ID: 0db3670b-4c12-1234-51b7-021923026132. CSeq: 87299500 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. X-AUTH-IP: x.x.x.174. P-Accountcode: . X-FS-Support: update_display,send_info. Remote-Party-ID: "xxxxxxx00181" ;party=calling;screen=yes;privacy=off. Content-Type: application/sdp. Content-Length: 261. . v=0. o=FreeSWITCH 1455243574 1455243575 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 27554 RTP/AVP 8 0 9 3 98 18 101 13. a=rtpmap:98 G7221/16000. a=fmtp:98 bitrate=32000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U x.x.x.166:5060 -> x.x.x.3:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0. Record-Route: . From: "xxxxxxx00181" ;tag=9pp9134ya08pe. To: . Call-ID: 0db3670b-4c12-1234-51b7-021923026132. CSeq: 87299500 INVITE. User-Agent: M01. Content-Length: 0. . U x.x.x.166:5060 -> x.x.x.3:5060 SIP/2.0 603 Decline. Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0. Max-Forwards: 25. From: "xxxxxxx00181" ;tag=9pp9134ya08pe. To: ;tag=vj0H5NyryajBm. Call-ID: 0db3670b-4c12-1234-51b7-021923026132. CSeq: 87299500 INVITE. User-Agent: M01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Reason: Q.850;cause=21;text="CALL_REJECTED". Content-Length: 0. . U x.x.x.3:5060 -> x.x.x.166:5060 ACK sip:xxxxxxx29858 at x.x.x.166 SIP/2.0. Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK93e3.c07e36f3.0. From: "xxxxxxx00181" ;tag=9pp9134ya08pe. Call-ID: 0db3670b-4c12-1234-51b7-021923026132. To: ;tag=vj0H5NyryajBm. CSeq: 87299500 ACK. Max-Forwards: 70. Content-Length: 0. . U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK789FQF04S4yte. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "xxxxxxx29858" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313. Record-Route: . From: ;tag=1037646889. To: ;tag=t0D01ZvH4r55c. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 INVITE. Contact: . User-Agent: M01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. . v=0. o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166. s=FreeSWITCH. c=IN IP4 x.x.x.166. t=0 0. m=audio 27468 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK789FQF04S4yte. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.166:5060 -> x.x.x.174:5060 ACK sip:xxxxxxx29858 at x.x.x.3:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK8H38raH8pDNDa. Route: . Route: . Max-Forwards: 70. From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 ACK. Contact: . Content-Length: 0. . U x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313. Record-Route: . From: ;tag=1037646889. To: ;tag=t0D01ZvH4r55c. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. . v=0. o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166. s=FreeSWITCH. c=IN IP4 x.x.x.166. t=0 0. m=audio 27468 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.0. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1418332271;rport=2313. Record-Route: . From: ;tag=1037646889. To: ;tag=t0D01ZvH4r55c. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. . v=0. o=FreeSWITCH 1455243571 1455243572 IN IP4 x.x.x.166. s=FreeSWITCH. c=IN IP4 x.x.x.166. t=0 0. m=audio 27468 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U x.x.x.174:5060 -> x.x.x.166:5060 ACK sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.2. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1857959530;rport=2313. From: ;tag=1037646889. To: ;tag=t0D01ZvH4r55c. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 ACK. Contact: . Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b. Max-Forwards: 30. Supported: replaces, path, timer. User-Agent: Grandstream GXP1400 1.0.6.11. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. P-hint: rr-enforced. X-AUTH-IP: x.x.x.47. . U x.x.x.174:5060 -> x.x.x.166:5060 ACK sip:xxxxxxx29858 at x.x.x.166:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bK2bf1.30ee31f1.2. Via: SIP/2.0/UDP x.x.x.47:2313;received=x.x.x.47;branch=z9hG4bK1857959530;rport=2313. From: ;tag=1037646889. To: ;tag=t0D01ZvH4r55c. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 71 ACK. Contact: . Proxy-Authorization: Digest username="xxxx9", realm="x.x.x.174", nonce="fd658a68-d16e-11e5-b0fa-b321f35ec85c", uri="sip:xxxxxxx29858 at x.x.x.174", response="97445fab50fed9e6930856844ec41daf", algorithm=MD5, cnonce="05356127", qop=auth, nc=0000000b. Max-Forwards: 30. Supported: replaces, path, timer. User-Agent: Grandstream GXP1400 1.0.6.11. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. P-hint: rr-enforced. X-AUTH-IP: x.x.x.47. . U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;rport=5060;received=x.x.x.166;branch=z9hG4bK8H38raH8pDNDa. Record-Route: . Record-Route: . From: "xxxxxxx00181" ;tag=U96r3tDN11Urr. To: ;tag=tFUN7D89Nttyg. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299453 INVITE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Allow-Events: talk, hold, conference, refer. Content-Disposition: session. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 242. . v=0. o=FreeSWITCH 1455252976 1455252977 IN IP4 x.x.x.6. s=FreeSWITCH. c=IN IP4 x.x.x.6. t=0 0. m=audio 18154 RTP/AVP 8 101 13. 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 x.x.x.174:5060 -> x.x.x.166:5060 BYE sip:gw+test_gateway at x.x.x.166:5060;transport=udp;gw=test_gateway SIP/2.0. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bKb4be.671066d2.0. Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bKb4be.fba2e636.0. Max-Forwards: 29. From: ;tag=tFUN7D89Nttyg. To: "xxxxxxx00181" ;tag=U96r3tDN11Urr. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299520 BYE. Contact: . User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Reason: SIP;cause=408;text="ACK Timeout". P-hint: rr-enforced. Content-Length: 0. P-hint: rr-enforced. X-AUTH-IP: x.x.x.3. . U x.x.x.166:5060 -> x.x.x.174:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.174:5060;branch=z9hG4bKb4be.671066d2.0. Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bKb4be.fba2e636.0. From: ;tag=tFUN7D89Nttyg. To: "xxxxxxx00181" ;tag=U96r3tDN11Urr. Call-ID: d5625619-4c11-1234-0fb6-40a8f0282f84. CSeq: 87299520 BYE. User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Content-Length: 0. . U x.x.x.166:5060 -> x.x.x.174:5060 BYE sip:xxxx9 at x.x.x.47:2313 SIP/2.0. Via: SIP/2.0/UDP x.x.x.166;rport;branch=z9hG4bK9tv1t51BmpB0N. Route: . Max-Forwards: 70. From: ;tag=t0D01ZvH4r55c. To: ;tag=1037646889. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 87299474 BYE. User-Agent: M01. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. P-hint: rr-enforced. X-AUTH-IP: x.x.x.3. . U x.x.x.174:5060 -> x.x.x.166:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP x.x.x.166;received=x.x.x.166;rport=5060;branch=z9hG4bK9tv1t51BmpB0N. From: ;tag=t0D01ZvH4r55c. To: ;tag=1037646889. Call-ID: 1770373286-5062-8 at BJC.BGI.B.BFJ. CSeq: 87299474 BYE. Contact: . Supported: replaces, path, timer. User-Agent: Grandstream GXP1400 1.0.6.11. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . From avi at avimarcus.net Mon Feb 22 09:36:14 2016 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 22 Feb 2016 06:36:14 +0000 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: References: Message-ID: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> 5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel wrote: > Hello, > > Having issue of call dropping after 32 seconds, here are the details- > > x.x.x.174: opensips server > x.x.x.166: freeswitch server > x.x.x.3: another opensips server which is registered as gateway on > above freeswitch server > x.x.x.6: freeswitch server > x.x.x.47: server through which the user is registered > I am trying to call from xxxx9 to xxxxxxx29858 > xxxxxxx00181 is caller-id name and caller-id number > > Call flow is like this: > registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips > server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) > > Outbound-proxy is set to x.x.x.174 in Gateway configuration. > > 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 > 2) Call hit the freeswitch server x.x.x.166 > 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving > started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 > But after 32 seconds call is dropped, > Within 32 seconds audio is ok from both end so it should not be the RTP > issue. > Here I have attached the file with sip logs, you can observer from the > file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 > and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then > 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. > > What is wrong here? Any help would be appreciated here. > > Here is the file with sip logs > -- > Thanks, > Rutu Patel > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160222/44cb330a/attachment.html From gascagonzalo at gmail.com Mon Feb 22 12:58:12 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Mon, 22 Feb 2016 01:58:12 -0800 Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8 Message-ID: Compiling latest FS version getting the following error when running make: You must install libvpx2-dev to build mod_av. Stop. *Using this guide:* https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video *Version: **Debian 8.3 x64* *Troubleshooting* Previous version (Last week) was working fine in 2 different systems. *git log* commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935 Author: Anthony Minessale Date: Fri Feb 19 09:34:38 2016 -0600 FS-8847 #resolve [Silence Error on shutdown of video call] commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad Author: Anthony Minessale Date: Thu Feb 18 17:04:01 2016 -0600 *git diff* https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995 *logs (make)* making all mod_amr make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' make all-am make[5]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' CC mod_amr_la-mod_amr.lo CCLD mod_amr.la make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' making all mod_av make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/applications/mod_av' Makefile:915: *** You must install libvpx2-dev to build mod_av. Stop. make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/applications/mod_av' Makefile:648: recipe for target 'mod_av-all' failed /usr/src/freeswitch.git/src/mod/applications/mod_av# tree . ??? avcodec.c ??? avformat.c ??? Makefile ??? Makefile.am ??? Makefile.in ??? mod_av.c 0 directories, 6 files root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# ls -alh total 208K drwxr-xr-x 3 root root 4.0K Feb 22 04:46 . drwxr-xr-x 59 root root 4.0K Feb 22 04:20 .. -rw-r--r-- 1 root root 46K Feb 22 04:20 avcodec.c -rw-r--r-- 1 root root 63K Feb 22 04:20 avformat.c drwxr-xr-x 2 root root 4.0K Feb 22 04:24 .deps -rw-r--r-- 1 root root 35K Feb 22 04:24 Makefile -rw-r--r-- 1 root root 828 Feb 22 04:20 Makefile.am -rw-r--r-- 1 root root 33K Feb 22 04:23 Makefile.in -rw-r--r-- 1 root root 4.3K Feb 22 04:20 mod_av.c *Found* http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html But not sure how to proceed for this release Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160222/83b5a29a/attachment.html From gmaruzz at gmail.com Mon Feb 22 14:12:28 2016 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 22 Feb 2016 12:12:28 +0100 Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8 In-Reply-To: References: Message-ID: As told in irc, you must install *unstable* repository for compiling latest git master. Check on www.freeswitch.conf/confluence the page for debian 8 jessie with updated instruction Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" ha scritto: > Compiling latest FS version getting the following error when running make: > > You must install libvpx2-dev to build mod_av. Stop. > > *Using this guide:* > > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video > > *Version: **Debian 8.3 x64* > > *Troubleshooting* > > Previous version (Last week) was working fine in 2 different systems. > > *git log* > > commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935 > > Author: Anthony Minessale > > Date: Fri Feb 19 09:34:38 2016 -0600 > > FS-8847 #resolve [Silence Error on shutdown of video call] > > commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad > > Author: Anthony Minessale > > Date: Thu Feb 18 17:04:01 2016 -0600 > > *git diff* > > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995 > > > *logs (make)* > > > making all mod_amr > > make[4]: Entering directory > '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' > > make all-am > > make[5]: Entering directory > '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' > > CC mod_amr_la-mod_amr.lo > > CCLD mod_amr.la > > make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' > > make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' > > making all mod_av > > make[4]: Entering directory > '/usr/src/freeswitch.git/src/mod/applications/mod_av' > > Makefile:915: *** You must install libvpx2-dev to build mod_av. Stop. > > make[4]: Leaving directory > '/usr/src/freeswitch.git/src/mod/applications/mod_av' > > Makefile:648: recipe for target 'mod_av-all' failed > > > /usr/src/freeswitch.git/src/mod/applications/mod_av# tree > > . > > ??? avcodec.c > > ??? avformat.c > > ??? Makefile > > ??? Makefile.am > > ??? Makefile.in > > ??? mod_av.c > > > 0 directories, 6 files > > root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# > ls -alh > > total 208K > > drwxr-xr-x 3 root root 4.0K Feb 22 04:46 . > > drwxr-xr-x 59 root root 4.0K Feb 22 04:20 .. > > -rw-r--r-- 1 root root 46K Feb 22 04:20 avcodec.c > > -rw-r--r-- 1 root root 63K Feb 22 04:20 avformat.c > > drwxr-xr-x 2 root root 4.0K Feb 22 04:24 .deps > > -rw-r--r-- 1 root root 35K Feb 22 04:24 Makefile > > -rw-r--r-- 1 root root 828 Feb 22 04:20 Makefile.am > > -rw-r--r-- 1 root root 33K Feb 22 04:23 Makefile.in > > -rw-r--r-- 1 root root 4.3K Feb 22 04:20 mod_av.c > > *Found* > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html > But not sure how to proceed for this release > > 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/20160222/f6c6a1ca/attachment-0001.html From aubalde at presenceco.com Mon Feb 22 18:51:13 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Mon, 22 Feb 2016 16:51:13 +0100 Subject: [Freeswitch-users] Invalid sdp on WebRTC connections Message-ID: We have a WebRTC environment with Freeswitch and sipml5. The fact is that when a WebRTC extension receives a call, the SDP that sends Freeswitch not show the correct candidate: *======================================================================* *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:* *22d33210-d709-11e5-bdcd-078a69e08f30 v=0* *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973 IN IP4 X.X.X.X* *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH* *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X* *======================================================================* Where X.X.X.X is the internal IP of Freeswitch beghind the NAT. Any idea? Thanks, *PRESENCE TECHNOLOGY* *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* *Follow us on:* *[image: tw]* *[image: yt]* *[image: in]* *[image: ss]* *[image: fb]* For additional information, please visit our website *www.presenceco.com* -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160222/85381fb7/attachment.html From mgiammarco at gmail.com Mon Feb 22 20:31:28 2016 From: mgiammarco at gmail.com (Mario Giammarco) Date: Mon, 22 Feb 2016 17:31:28 +0000 (UTC) Subject: [Freeswitch-users] Recording and mono directional webrtc video Message-ID: Hello, I am evaluating freeswitch for this purpose: I need a bidirectional audio call with one direction video only from caller. And I need to record audio and video from caller. I need to use webrtc protocol. I have seen old threads that show that FreeSwitch has support for these things but I would like to be sure that the implementation is now on main tree and it is really supported. Thanks, Mario From anthony.minessale at gmail.com Mon Feb 22 22:15:48 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 22 Feb 2016 13:15:48 -0600 Subject: [Freeswitch-users] Invalid sdp on WebRTC connections In-Reply-To: References: Message-ID: See ext-rtp-ip and apply-candidate-acl On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde wrote: > We have a WebRTC environment with Freeswitch and sipml5. The fact is that > when a WebRTC extension receives a call, the SDP that sends Freeswitch not > show the correct candidate: > *======================================================================* > *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:* > *22d33210-d709-11e5-bdcd-078a69e08f30 v=0* > *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973 > IN IP4 X.X.X.X* > *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH* > *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X* > *======================================================================* > > Where X.X.X.X is the internal IP of Freeswitch beghind the NAT. > Any idea? > > > Thanks, > > *PRESENCE TECHNOLOGY* > *Agust? Ubalde Bellot* > Chief Developer > C/ Comte Urgell 240 3A > Barcelona 08036 > aubalde at presenceco.com > > Ph: +34 93 10 10 300 > Fx: +34 93 10 10 333 > > *www.presenceco.com* > > *Follow us on:* > > *[image: tw]* *[image: yt]* > *[image: in]* > *[image: ss]* > *[image: fb]* > > > For additional information, please visit our website *www.presenceco.com* > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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 > -- 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/20160222/949718f0/attachment-0001.html From gascagonzalo at gmail.com Tue Feb 23 03:16:22 2016 From: gascagonzalo at gmail.com (Gonzalo Gasca Meza) Date: Mon, 22 Feb 2016 16:16:22 -0800 Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8 In-Reply-To: References: Message-ID: Just as a note Recent builds are failing with similar error: https://freeswitch.org/bamboo/browse/FS-TMB/latest On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli wrote: > As told in irc, you must install *unstable* repository for compiling > latest git master. > > Check on www.freeswitch.conf/confluence the page for debian 8 jessie with > updated instruction > Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" ha > scritto: > >> Compiling latest FS version getting the following error when running make: >> >> You must install libvpx2-dev to build mod_av. Stop. >> >> *Using this guide:* >> >> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >> >> *Version: **Debian 8.3 x64* >> >> *Troubleshooting* >> >> Previous version (Last week) was working fine in 2 different systems. >> >> *git log* >> >> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935 >> >> Author: Anthony Minessale >> >> Date: Fri Feb 19 09:34:38 2016 -0600 >> >> FS-8847 #resolve [Silence Error on shutdown of video call] >> >> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad >> >> Author: Anthony Minessale >> >> Date: Thu Feb 18 17:04:01 2016 -0600 >> >> *git diff* >> >> >> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995 >> >> >> *logs (make)* >> >> >> making all mod_amr >> >> make[4]: Entering directory >> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >> >> make all-am >> >> make[5]: Entering directory >> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >> >> CC mod_amr_la-mod_amr.lo >> >> CCLD mod_amr.la >> >> make[5]: Leaving directory >> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >> >> make[4]: Leaving directory >> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >> >> making all mod_av >> >> make[4]: Entering directory >> '/usr/src/freeswitch.git/src/mod/applications/mod_av' >> >> Makefile:915: *** You must install libvpx2-dev to build mod_av. Stop. >> >> make[4]: Leaving directory >> '/usr/src/freeswitch.git/src/mod/applications/mod_av' >> >> Makefile:648: recipe for target 'mod_av-all' failed >> >> >> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree >> >> . >> >> ??? avcodec.c >> >> ??? avformat.c >> >> ??? Makefile >> >> ??? Makefile.am >> >> ??? Makefile.in >> >> ??? mod_av.c >> >> >> 0 directories, 6 files >> >> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# >> ls -alh >> >> total 208K >> >> drwxr-xr-x 3 root root 4.0K Feb 22 04:46 . >> >> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 .. >> >> -rw-r--r-- 1 root root 46K Feb 22 04:20 avcodec.c >> >> -rw-r--r-- 1 root root 63K Feb 22 04:20 avformat.c >> >> drwxr-xr-x 2 root root 4.0K Feb 22 04:24 .deps >> >> -rw-r--r-- 1 root root 35K Feb 22 04:24 Makefile >> >> -rw-r--r-- 1 root root 828 Feb 22 04:20 Makefile.am >> >> -rw-r--r-- 1 root root 33K Feb 22 04:23 Makefile.in >> >> -rw-r--r-- 1 root root 4.3K Feb 22 04:20 mod_av.c >> >> *Found* >> >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html >> But not sure how to proceed for this release >> >> 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160222/8e55a5a9/attachment.html From krice at freeswitch.org Tue Feb 23 03:51:36 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 22 Feb 2016 18:51:36 -0600 Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8 In-Reply-To: References: Message-ID: <3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org> This is a known issue as this deb 7 build node is about to be replaced and will not be fixed. It just needs an upgraded libvpx2 there will be failures off and on like this in he short term as chrome is in the process of upgrading this lib and unfortunately its forcing us to do the same. This is the master branch so it can be broken at times. Sent from my iPhone > On Feb 22, 2016, at 6:16 PM, Gonzalo Gasca Meza wrote: > > Just as a note > Recent builds are failing with similar error: > > https://freeswitch.org/bamboo/browse/FS-TMB/latest > > >> On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli wrote: >> As told in irc, you must install *unstable* repository for compiling latest git master. >> >> Check on www.freeswitch.conf/confluence the page for debian 8 jessie with updated instruction >> >> Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" ha scritto: >>> Compiling latest FS version getting the following error when running make: >>> >>> You must install libvpx2-dev to build mod_av. Stop. >>> >>> Using this guide: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >>> >>> Version: Debian 8.3 x64 >>> >>> Troubleshooting >>> >>> Previous version (Last week) was working fine in 2 different systems. >>> >>> git log >>> >>> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935 >>> >>> Author: Anthony Minessale >>> >>> Date: Fri Feb 19 09:34:38 2016 -0600 >>> >>> FS-8847 #resolve [Silence Error on shutdown of video call] >>> >>> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad >>> >>> Author: Anthony Minessale >>> >>> >>> Date: Thu Feb 18 17:04:01 2016 -0600 >>> >>> git diff >>> >>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995 >>> >>> >>> >>> logs (make) >>> >>> >>> >>> making all mod_amr >>> >>> make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> make all-am >>> >>> make[5]: Entering directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> CC mod_amr_la-mod_amr.lo >>> >>> CCLD mod_amr.la >>> >>> make[5]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> making all mod_av >>> >>> make[4]: Entering directory '/usr/src/freeswitch.git/src/mod/applications/mod_av' >>> >>> Makefile:915: *** You must install libvpx2-dev to build mod_av. Stop. >>> >>> make[4]: Leaving directory '/usr/src/freeswitch.git/src/mod/applications/mod_av' >>> >>> Makefile:648: recipe for target 'mod_av-all' failed >>> >>> >>> >>> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree >>> >>> . >>> >>> ??? avcodec.c >>> >>> ??? avformat.c >>> >>> ??? Makefile >>> >>> ??? Makefile.am >>> >>> ??? Makefile.in >>> >>> ??? mod_av.c >>> >>> >>> >>> 0 directories, 6 files >>> >>> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# ls -alh >>> >>> total 208K >>> >>> drwxr-xr-x 3 root root 4.0K Feb 22 04:46 . >>> >>> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 .. >>> >>> -rw-r--r-- 1 root root 46K Feb 22 04:20 avcodec.c >>> >>> -rw-r--r-- 1 root root 63K Feb 22 04:20 avformat.c >>> >>> drwxr-xr-x 2 root root 4.0K Feb 22 04:24 .deps >>> >>> -rw-r--r-- 1 root root 35K Feb 22 04:24 Makefile >>> >>> -rw-r--r-- 1 root root 828 Feb 22 04:20 Makefile.am >>> >>> -rw-r--r-- 1 root root 33K Feb 22 04:23 Makefile.in >>> >>> >>> -rw-r--r-- 1 root root 4.3K Feb 22 04:20 mod_av.c >>> >>> Found >>> >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html >>> >>> But not sure how to proceed for this release >>> >>> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160222/2c208602/attachment-0001.html From s.safarov at gmail.com Tue Feb 23 06:15:14 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 23 Feb 2016 03:15:14 +0000 Subject: [Freeswitch-users] Freeswitch libvpx2-dev compile error Debian 8 In-Reply-To: <3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org> References: <3CAF63BC-F20D-4CA8-A5B0-3E091BC7157A@freeswitch.org> Message-ID: Please check version of instaled libvpx2. Must be 1.5 according SD repo. On Tue, Feb 23, 2016, 03:52 Ken Rice wrote: > This is a known issue as this deb 7 build node is about to be replaced and > will not be fixed. It just needs an upgraded libvpx2 there will be failures > off and on like this in he short term as chrome is in the process of > upgrading this lib and unfortunately its forcing us to do the same. This is > the master branch so it can be broken at times. > > Sent from my iPhone > > On Feb 22, 2016, at 6:16 PM, Gonzalo Gasca Meza > wrote: > > Just as a note > Recent builds are failing with similar error: > > https://freeswitch.org/bamboo/browse/FS-TMB/latest > > > On Mon, Feb 22, 2016 at 3:12 AM, Giovanni Maruzzelli > wrote: > >> As told in irc, you must install *unstable* repository for compiling >> latest git master. >> >> Check on www.freeswitch.conf/confluence the page for debian 8 jessie >> with updated instruction >> Il 22/Feb/2016 10:59, "Gonzalo Gasca Meza" ha >> scritto: >> >>> Compiling latest FS version getting the following error when running >>> make: >>> >>> You must install libvpx2-dev to build mod_av. Stop. >>> >>> *Using this guide:* >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video >>> >>> *Version: **Debian 8.3 x64* >>> >>> *Troubleshooting* >>> >>> Previous version (Last week) was working fine in 2 different systems. >>> >>> *git log* >>> >>> commit 3bd26eaa6b2dff1d80cac4e58fc860fb96f92935 >>> >>> Author: Anthony Minessale >>> >>> Date: Fri Feb 19 09:34:38 2016 -0600 >>> >>> FS-8847 #resolve [Silence Error on shutdown of video call] >>> >>> commit 9f4f67df5d2d9606ba88023b1bd05d7b88c69dad >>> >>> Author: Anthony Minessale >>> >>> Date: Thu Feb 18 17:04:01 2016 -0600 >>> >>> *git diff* >>> >>> >>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_av/Makefile.am?until=4ad0aa91a4a34fd10782dfe21aba74d49724f995 >>> >>> >>> *logs (make)* >>> >>> >>> making all mod_amr >>> >>> make[4]: Entering directory >>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> make all-am >>> >>> make[5]: Entering directory >>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> CC mod_amr_la-mod_amr.lo >>> >>> CCLD mod_amr.la >>> >>> make[5]: Leaving directory >>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> make[4]: Leaving directory >>> '/usr/src/freeswitch.git/src/mod/codecs/mod_amr' >>> >>> making all mod_av >>> >>> make[4]: Entering directory >>> '/usr/src/freeswitch.git/src/mod/applications/mod_av' >>> >>> Makefile:915: *** You must install libvpx2-dev to build mod_av. Stop. >>> >>> make[4]: Leaving directory >>> '/usr/src/freeswitch.git/src/mod/applications/mod_av' >>> >>> Makefile:648: recipe for target 'mod_av-all' failed >>> >>> >>> /usr/src/freeswitch.git/src/mod/applications/mod_av# tree >>> >>> . >>> >>> ??? avcodec.c >>> >>> ??? avformat.c >>> >>> ??? Makefile >>> >>> ??? Makefile.am >>> >>> ??? Makefile.in >>> >>> ??? mod_av.c >>> >>> >>> 0 directories, 6 files >>> >>> root at nkn-freeswitch-sfo1:/usr/src/freeswitch.git/src/mod/applications/mod_av# >>> ls -alh >>> >>> total 208K >>> >>> drwxr-xr-x 3 root root 4.0K Feb 22 04:46 . >>> >>> drwxr-xr-x 59 root root 4.0K Feb 22 04:20 .. >>> >>> -rw-r--r-- 1 root root 46K Feb 22 04:20 avcodec.c >>> >>> -rw-r--r-- 1 root root 63K Feb 22 04:20 avformat.c >>> >>> drwxr-xr-x 2 root root 4.0K Feb 22 04:24 .deps >>> >>> -rw-r--r-- 1 root root 35K Feb 22 04:24 Makefile >>> >>> -rw-r--r-- 1 root root 828 Feb 22 04:20 Makefile.am >>> >>> -rw-r--r-- 1 root root 33K Feb 22 04:23 Makefile.in >>> >>> -rw-r--r-- 1 root root 4.3K Feb 22 04:20 mod_av.c >>> >>> *Found* >>> >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116401.html >>> But not sure how to proceed for this release >>> >>> 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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160223/aecabe76/attachment.html From nandy1925 at gmail.com Tue Feb 23 06:23:16 2016 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Tue, 23 Feb 2016 11:23:16 +0800 Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION Message-ID: Hello guys, I installed mod_gsmopen. When testing incoming call, it hangs up with this error message INCOMPATIBLE_DESTINATION. I understand this is a codec negotiation issue. Looking at the CDR - the gsmopen presented L16 8000 in the SDP. By default, FS does early negotiation. Snippet of my internal sip_profile: Take note: there is no disable-transcoding parameter which I understand FS is transcodind. In vars.xml: I tried to add L16 at the end of global_codec_prefs which is unnecessary, isn't it? But it didn't work, too. Finally, I added L16 in the softphone, it worked. I have installed gsmopen before and it worked (transcoding) with IP phones. Is there anything I may have missed? I appreciate for any leads. Thank you, /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/df7768b7/attachment-0001.html From rutu.patel at inextrix.com Tue Feb 23 08:51:55 2016 From: rutu.patel at inextrix.com (Rutu Patel) Date: Tue, 23 Feb 2016 11:21:55 +0530 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> References: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> Message-ID: Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? >From where can i start? where to look? Thanks. -- Thanks, Rutu Patel On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: > 5 second response: 32 seconds is a timer/[network/NAT] issue. > > You have lots of 200s to the user since it's waiting for an ACK and keeps > retrying, but for whatever network reason (router... sip alg?), it isn't > getting one, so it triggers a timer to stop the call. > > > -Avi Marcus > BestFone > > On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel > wrote: > >> Hello, >> >> Having issue of call dropping after 32 seconds, here are the details- >> >> x.x.x.174: opensips server >> x.x.x.166: freeswitch server >> x.x.x.3: another opensips server which is registered as gateway on >> above freeswitch server >> x.x.x.6: freeswitch server >> x.x.x.47: server through which the user is registered >> I am trying to call from xxxx9 to xxxxxxx29858 >> xxxxxxx00181 is caller-id name and caller-id number >> >> Call flow is like this: >> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips >> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >> >> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >> >> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >> 2) Call hit the freeswitch server x.x.x.166 >> 3) After '180 Ringing' and '183 Session Progress' packet >> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >> gateway x.x.x.3 >> But after 32 seconds call is dropped, >> Within 32 seconds audio is ok from both end so it should not be the RTP >> issue. >> Here I have attached the file with sip logs, you can observer from the >> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >> >> What is wrong here? Any help would be appreciated here. >> >> Here is the file with sip logs >> -- >> Thanks, >> Rutu Patel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160223/fd762723/attachment.html From miha at softnet.si Tue Feb 23 10:11:29 2016 From: miha at softnet.si (Miha) Date: Tue, 23 Feb 2016 08:11:29 +0100 Subject: [Freeswitch-users] Freeswitch Centos 7 installation Message-ID: <56CC0621.70607@softnet.si> Hi, i am having some problem with FS installation on Centos 7. Here a log from "make": making all mod_v8 make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8' cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || patch -t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build. patch) /bin/sh: patch: command not found make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127 make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8' make[3]: *** [mod_v8-all] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [all] Error 2 Could some point me on right direction to get is working? tnx miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/fda349c2/attachment.html From krice at freeswitch.org Tue Feb 23 10:19:08 2016 From: krice at freeswitch.org (Ken Rice) Date: Tue, 23 Feb 2016 01:19:08 -0600 Subject: [Freeswitch-users] Freeswitch Centos 7 installation In-Reply-To: <56CC0621.70607@softnet.si> References: <56CC0621.70607@softnet.si> Message-ID: <298201d16e0a$7cb24040$7616c0c0$@freeswitch.org> The line ?/bin/sh: patch: command not found? is a big hint you might need to make sure you have patch installed From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha Sent: Tuesday, February 23, 2016 1:11 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Freeswitch Centos 7 installation Hi, i am having some problem with FS installation on Centos 7. Here a log from "make": making all mod_v8 make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8' cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || patch -t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build. patch) /bin/sh: patch: command not found make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127 make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8' make[3]: *** [mod_v8-all] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [all] Error 2 Could some point me on right direction to get is working? tnx miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/179295c5/attachment.html From jurij.ivo at gmail.com Tue Feb 23 10:37:39 2016 From: jurij.ivo at gmail.com (Jurijs Ivolga) Date: Tue, 23 Feb 2016 09:37:39 +0200 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: References: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> Message-ID: Hi, 1) You have very complex set-up and I doubt that you need it. 2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP. 3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty. With kind regards, Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel : > Thanks for the reply. > > Got your point about NATing issue and no response of 200 OK and as a > resoult ACK Timeout. > So, now to resolve the issue, if you can assist, what could be the > possible fixies? > From where can i start? where to look? > > Thanks. > > -- > Thanks, > Rutu Patel > > > > On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: > >> 5 second response: 32 seconds is a timer/[network/NAT] issue. >> >> You have lots of 200s to the user since it's waiting for an ACK and keeps >> retrying, but for whatever network reason (router... sip alg?), it isn't >> getting one, so it triggers a timer to stop the call. >> >> >> -Avi Marcus >> BestFone >> >> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >> wrote: >> >>> Hello, >>> >>> Having issue of call dropping after 32 seconds, here are the details- >>> >>> x.x.x.174: opensips server >>> x.x.x.166: freeswitch server >>> x.x.x.3: another opensips server which is registered as gateway on >>> above freeswitch server >>> x.x.x.6: freeswitch server >>> x.x.x.47: server through which the user is registered >>> I am trying to call from xxxx9 to xxxxxxx29858 >>> xxxxxxx00181 is caller-id name and caller-id number >>> >>> Call flow is like this: >>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips >>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>> >>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>> >>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>> 2) Call hit the freeswitch server x.x.x.166 >>> 3) After '180 Ringing' and '183 Session Progress' packet >>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>> gateway x.x.x.3 >>> But after 32 seconds call is dropped, >>> Within 32 seconds audio is ok from both end so it should not be the RTP >>> issue. >>> Here I have attached the file with sip logs, you can observer from the >>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>> >>> What is wrong here? Any help would be appreciated here. >>> >>> Here is the file with sip logs >>> -- >>> Thanks, >>> Rutu Patel >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 > Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel : > Thanks for the reply. > > Got your point about NATing issue and no response of 200 OK and as a > resoult ACK Timeout. > So, now to resolve the issue, if you can assist, what could be the > possible fixies? > From where can i start? where to look? > > Thanks. > > -- > Thanks, > Rutu Patel > > > > On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: > >> 5 second response: 32 seconds is a timer/[network/NAT] issue. >> >> You have lots of 200s to the user since it's waiting for an ACK and keeps >> retrying, but for whatever network reason (router... sip alg?), it isn't >> getting one, so it triggers a timer to stop the call. >> >> >> -Avi Marcus >> BestFone >> >> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >> wrote: >> >>> Hello, >>> >>> Having issue of call dropping after 32 seconds, here are the details- >>> >>> x.x.x.174: opensips server >>> x.x.x.166: freeswitch server >>> x.x.x.3: another opensips server which is registered as gateway on >>> above freeswitch server >>> x.x.x.6: freeswitch server >>> x.x.x.47: server through which the user is registered >>> I am trying to call from xxxx9 to xxxxxxx29858 >>> xxxxxxx00181 is caller-id name and caller-id number >>> >>> Call flow is like this: >>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips >>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>> >>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>> >>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>> 2) Call hit the freeswitch server x.x.x.166 >>> 3) After '180 Ringing' and '183 Session Progress' packet >>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>> gateway x.x.x.3 >>> But after 32 seconds call is dropped, >>> Within 32 seconds audio is ok from both end so it should not be the RTP >>> issue. >>> Here I have attached the file with sip logs, you can observer from the >>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>> >>> What is wrong here? Any help would be appreciated here. >>> >>> Here is the file with sip logs >>> -- >>> Thanks, >>> Rutu Patel >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160223/c6e2127f/attachment-0001.html From lte at lte-net.de Tue Feb 23 11:40:16 2016 From: lte at lte-net.de (Fred Schulz) Date: Tue, 23 Feb 2016 09:40:16 +0100 Subject: [Freeswitch-users] mod_com_g729 validator Message-ID: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> Hi all, does one know if it is possible to pass parameters towards the validator? We plan to install not just only one system, so we?re trying to use a script. Therefore it would be easier to install our systems. Thanks Fred From miha at softnet.si Tue Feb 23 11:49:34 2016 From: miha at softnet.si (Miha) Date: Tue, 23 Feb 2016 09:49:34 +0100 Subject: [Freeswitch-users] Freeswitch Centos 7 installation In-Reply-To: <298201d16e0a$7cb24040$7616c0c0$@freeswitch.org> References: <56CC0621.70607@softnet.si> <298201d16e0a$7cb24040$7616c0c0$@freeswitch.org> Message-ID: <56CC1D1E.6040504@softnet.si> Tnx Ken:) I should i known that :) br miha On 23/02/2016 08:19, Ken Rice wrote: > > The line ?/bin/sh: patch: command not found? is a big hint you might > need to make sure you have patch installed > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Miha > *Sent:* Tuesday, February 23, 2016 1:11 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Freeswitch Centos 7 installation > > Hi, > > i am having some problem with FS installation on Centos 7. > > Here a log from "make": > making all mod_v8 > make[4]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8' > cd /usr/src/freeswitch/libs/v8-3.24.14 && (test -f .stamp-patch || > patch -t -p0 < /usr/src/freeswitch/src/mod/languages/mod_v8/v8-build. > patch) > /bin/sh: patch: command not found > make[4]: *** [/usr/src/freeswitch/libs/v8-3.24.14/.stamp-patch] Error 127 > make[4]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8' > make[3]: *** [mod_v8-all] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [all] Error 2 > > Could some point me on right direction to get is working? > tnx > miha > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/78d27704/attachment.html From aqsyounas at gmail.com Tue Feb 23 16:34:22 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 23 Feb 2016 18:34:22 +0500 Subject: [Freeswitch-users] Unable to detect beep with mod_avmd Message-ID: Hi, I am using avmd to detect beep over a channel. Beep is generated by below script. and starts avmd on fs_cli as. freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start I could proper hear beeps over the phone but does not see avmd detecting this. Does not see anything like "beep detected" in logs. I am made several attempts but unable to detect beep even a single time. Is there something wrong that i am doing? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/3f674a1f/attachment.html From aubalde at presenceco.com Tue Feb 23 11:15:41 2016 From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=) Date: Tue, 23 Feb 2016 09:15:41 +0100 Subject: [Freeswitch-users] Invalid sdp on WebRTC connections In-Reply-To: References: Message-ID: Hi, *ext-rtp-ip* is our public ip address. *apply-candidate-acl* is not defined at internal.xml. Thanks! *PRESENCE TECHNOLOGY* *Agust? Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde at presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* *Follow us on:* *[image: tw]* *[image: yt]* *[image: in]* *[image: ss]* *[image: fb]* For additional information, please visit our website *www.presenceco.com* 2016-02-22 20:15 GMT+01:00 Anthony Minessale : > See > > ext-rtp-ip and > > apply-candidate-acl > > > > On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde > wrote: > >> We have a WebRTC environment with Freeswitch and sipml5. The fact is that >> when a WebRTC extension receives a call, the SDP that sends Freeswitch not >> show the correct candidate: >> *======================================================================* >> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:* >> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0* >> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973 >> IN IP4 X.X.X.X* >> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH* >> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X* >> *======================================================================* >> >> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT. >> Any idea? >> >> >> Thanks, >> >> *PRESENCE TECHNOLOGY* >> *Agust? Ubalde Bellot* >> Chief Developer >> C/ Comte Urgell 240 3A >> Barcelona 08036 >> aubalde at presenceco.com >> >> Ph: +34 93 10 10 300 >> Fx: +34 93 10 10 333 >> >> *www.presenceco.com* >> >> *Follow us on:* >> >> *[image: tw]* *[image: yt]* >> *[image: in]* >> *[image: ss]* >> *[image: fb]* >> >> >> For additional information, please visit our website *www.presenceco.com* >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 >> > > > > -- > 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 > -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/c77d225a/attachment-0001.html From brian at freeswitch.org Tue Feb 23 17:44:01 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Feb 2016 08:44:01 -0600 Subject: [Freeswitch-users] Invalid sdp on WebRTC connections In-Reply-To: References: Message-ID: I think he's stating that you need to define apply-candidate-acl On Tue, Feb 23, 2016 at 2:15 AM, Agust? Ubalde wrote: > Hi, > > *ext-rtp-ip* is our public ip address. > *apply-candidate-acl* is not defined at internal.xml. > > > Thanks! > > *PRESENCE TECHNOLOGY* > *Agust? Ubalde Bellot* > Chief Developer > C/ Comte Urgell 240 3A > Barcelona 08036 > aubalde at presenceco.com > > Ph: +34 93 10 10 300 > Fx: +34 93 10 10 333 > > *www.presenceco.com* > > *Follow us on:* > > *[image: tw]* *[image: yt]* > *[image: in]* > *[image: ss]* > *[image: fb]* > > > For additional information, please visit our website *www.presenceco.com* > > > 2016-02-22 20:15 GMT+01:00 Anthony Minessale > : > >> See >> >> ext-rtp-ip and >> >> apply-candidate-acl >> >> >> >> On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde >> wrote: >> >>> We have a WebRTC environment with Freeswitch and sipml5. The fact is >>> that when a WebRTC extension receives a call, the SDP that sends Freeswitch >>> not show the correct candidate: >>> *======================================================================* >>> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:* >>> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0* >>> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 1455870973 >>> IN IP4 X.X.X.X* >>> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH* >>> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X* >>> *======================================================================* >>> >>> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT. >>> Any idea? >>> >>> >>> Thanks, >>> >>> *PRESENCE TECHNOLOGY* >>> *Agust? Ubalde Bellot* >>> Chief Developer >>> C/ Comte Urgell 240 3A >>> Barcelona 08036 >>> aubalde at presenceco.com >>> >>> Ph: +34 93 10 10 300 >>> Fx: +34 93 10 10 333 >>> >>> *www.presenceco.com* >>> >>> *Follow us on:* >>> >>> *[image: tw]* *[image: yt]* >>> *[image: in]* >>> *[image: ss]* >>> *[image: fb]* >>> >>> >>> For additional information, please visit our website >>> *www.presenceco.com* >>> >>> >>> *Presence Technology - DisclaimerThis message, its content and any file >>> attached thereto is for the intended recipient only and is confidential and >>> /or privileged. If you have received this e-mail in error or had access to >>> it, you should note that the information in it is private and any use >>> thereof is unauthorized. In such an event please notify us by e-mail or by >>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>> whatsoever means and any transmission or dissemination thereof to other >>> persons is prohibited. It should be deleted immediately from your system. >>> Presence Technology reserves the right to take legal action against any >>> persons unlawfully gaining access to the content of any external message it >>> has emitted.* >>> >>> *For additional information, please visit our website **www.presenceco.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 >>> >> >> >> >> -- >> 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 >> > > > > *Presence Technology - DisclaimerThis message, its content and any file > attached thereto is for the intended recipient only and is confidential and > /or privileged. If you have received this e-mail in error or had access to > it, you should note that the information in it is private and any use > thereof is unauthorized. In such an event please notify us by e-mail or by > telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by > whatsoever means and any transmission or dissemination thereof to other > persons is prohibited. It should be deleted immediately from your system. > Presence Technology reserves the right to take legal action against any > persons unlawfully gaining access to the content of any external message it > has emitted.* > > *For additional information, please visit our website **www.presenceco.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/20160223/aed12c8d/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 23 19:18:24 2016 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Feb 2016 10:18:24 -0600 Subject: [Freeswitch-users] Invalid sdp on WebRTC connections In-Reply-To: References: Message-ID: apply-candidate-acl controls the networks the candidates will be preferred from. They are parsed in order they are defined. On Tue, Feb 23, 2016 at 8:44 AM, Brian West wrote: > I think he's stating that you need to define apply-candidate-acl > > On Tue, Feb 23, 2016 at 2:15 AM, Agust? Ubalde > wrote: > >> Hi, >> >> *ext-rtp-ip* is our public ip address. >> *apply-candidate-acl* is not defined at internal.xml. >> >> >> Thanks! >> >> *PRESENCE TECHNOLOGY* >> *Agust? Ubalde Bellot* >> Chief Developer >> C/ Comte Urgell 240 3A >> Barcelona 08036 >> aubalde at presenceco.com >> >> Ph: +34 93 10 10 300 >> Fx: +34 93 10 10 333 >> >> *www.presenceco.com* >> >> *Follow us on:* >> >> *[image: tw]* *[image: yt]* >> *[image: in]* >> *[image: ss]* >> *[image: fb]* >> >> >> For additional information, please visit our website *www.presenceco.com* >> >> >> 2016-02-22 20:15 GMT+01:00 Anthony Minessale > >: >> >>> See >>> >>> ext-rtp-ip and >>> >>> apply-candidate-acl >>> >>> >>> >>> On Mon, Feb 22, 2016 at 9:51 AM, Agust? Ubalde >>> wrote: >>> >>>> We have a WebRTC environment with Freeswitch and sipml5. The fact is >>>> that when a WebRTC extension receives a call, the SDP that sends Freeswitch >>>> not show the correct candidate: >>>> *======================================================================* >>>> *22d33210-d709-11e5-bdcd-078a69e08f30 Local SDP:* >>>> *22d33210-d709-11e5-bdcd-078a69e08f30 v=0* >>>> *22d33210-d709-11e5-bdcd-078a69e08f30 o=FreeSWITCH 1455870972 >>>> 1455870973 IN IP4 X.X.X.X* >>>> *22d33210-d709-11e5-bdcd-078a69e08f30 s=FreeSWITCH* >>>> *22d33210-d709-11e5-bdcd-078a69e08f30 c=IN IP4 X.X.X.X* >>>> *======================================================================* >>>> >>>> Where X.X.X.X is the internal IP of Freeswitch beghind the NAT. >>>> Any idea? >>>> >>>> >>>> Thanks, >>>> >>>> *PRESENCE TECHNOLOGY* >>>> *Agust? Ubalde Bellot* >>>> Chief Developer >>>> C/ Comte Urgell 240 3A >>>> Barcelona 08036 >>>> aubalde at presenceco.com >>>> >>>> Ph: +34 93 10 10 300 >>>> Fx: +34 93 10 10 333 >>>> >>>> *www.presenceco.com* >>>> >>>> *Follow us on:* >>>> >>>> *[image: tw]* *[image: yt]* >>>> *[image: in]* >>>> *[image: ss]* >>>> *[image: fb]* >>>> >>>> >>>> For additional information, please visit our website >>>> *www.presenceco.com* >>>> >>>> >>>> *Presence Technology - DisclaimerThis message, its content and any file >>>> attached thereto is for the intended recipient only and is confidential and >>>> /or privileged. If you have received this e-mail in error or had access to >>>> it, you should note that the information in it is private and any use >>>> thereof is unauthorized. In such an event please notify us by e-mail or by >>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >>>> whatsoever means and any transmission or dissemination thereof to other >>>> persons is prohibited. It should be deleted immediately from your system. >>>> Presence Technology reserves the right to take legal action against any >>>> persons unlawfully gaining access to the content of any external message it >>>> has emitted.* >>>> >>>> *For additional information, please visit our website **www.presenceco.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 >>>> >>> >>> >>> >>> -- >>> 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 >>> >> >> >> >> *Presence Technology - DisclaimerThis message, its content and any file >> attached thereto is for the intended recipient only and is confidential and >> /or privileged. If you have received this e-mail in error or had access to >> it, you should note that the information in it is private and any use >> thereof is unauthorized. In such an event please notify us by e-mail or by >> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by >> whatsoever means and any transmission or dissemination thereof to other >> persons is prohibited. It should be deleted immediately from your system. >> Presence Technology reserves the right to take legal action against any >> persons unlawfully gaining access to the content of any external message it >> has emitted.* >> >> *For additional information, please visit our website **www.presenceco.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 > -- 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/20160223/e0d9a9ab/attachment-0001.html From steveayre at gmail.com Tue Feb 23 19:38:16 2016 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 23 Feb 2016 16:38:16 +0000 Subject: [Freeswitch-users] mod_com_g729 validator In-Reply-To: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> Message-ID: You might be able to interact with it using expect On 23 February 2016 at 08:40, Fred Schulz wrote: > Hi all, > > does one know if it is possible to pass parameters towards the validator? > We plan to install not just only one system, so we?re trying to use a > script. Therefore it would be easier to install our systems. > > Thanks > Fred > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/8b4ace82/attachment.html From brian at freeswitch.org Tue Feb 23 20:00:40 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Feb 2016 11:00:40 -0600 Subject: [Freeswitch-users] mod_com_g729 validator In-Reply-To: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> Message-ID: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/g729_activate On Tue, Feb 23, 2016 at 2:40 AM, Fred Schulz wrote: > Hi all, > > does one know if it is possible to pass parameters towards the validator? > We plan to install not just only one system, so we?re trying to use a > script. Therefore it would be easier to install our systems. > > Thanks > Fred > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/1285bba4/attachment.html From lte at lte-net.de Tue Feb 23 21:36:01 2016 From: lte at lte-net.de (Fred Schulz) Date: Tue, 23 Feb 2016 19:36:01 +0100 Subject: [Freeswitch-users] mod_com_g729 validator In-Reply-To: References: <010F1A96-67F0-432A-AB90-79383BC86FB7@lte-net.de> Message-ID: <8A481BE2-D52A-412C-9D75-F4C6E69EEEAB@lte-net.de> Thanks I give it a try :) > Am 23.02.2016 um 18:00 schrieb Brian West : > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/perl/g729_activate > > > On Tue, Feb 23, 2016 at 2:40 AM, Fred Schulz > wrote: > Hi all, > > does one know if it is possible to pass parameters towards the validator? > We plan to install not just only one system, so we?re trying to use a script. Therefore it would be easier to install our systems. > > Thanks > Fred > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/0f3de447/attachment.html From victor.chukalovskiy at gmail.com Tue Feb 23 21:52:28 2016 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 23 Feb 2016 13:52:28 -0500 Subject: [Freeswitch-users] Incorrect reply to T.38 re-INVITE In-Reply-To: References: Message-ID: <56CCAA6C.3010901@gmail.com> Hi Tomasz, Hi, I was the guy who reported and followed-up on https://freeswitch.org/jira/browse/FS-6954 What I recall that it needed to be fixed for a variety of possible config call flow scenarios, that can be represented as 3 independent factors: Factor 1 - whether FS is in proxy_media, bypass_media, or default full media mode. (3 options) Factor 2 - direction in which t.38 reINVITE flows (A --> B or B --> A). (2 options) Factor 3 - whether A leg of the call and B leg of the call is the same SIP profile or two different SIP profiles. (2 options) So, you have a matrix representing possible scenarios based on these 3 independent factors...So, you have at least 3 x 2 x 2 = 12 distinct cases to test. Bugfixes were done for some of them, but there were some cases left where bug fix was never complete, at least not until I gave-up on that JIRA. I'd imagine cases that were fixed made it's way to 1.6, while cases that were not fixed can still be in the same state. If you want to be confident about current state, I'd recommend getting 1.6 and setting test environment where you can verify each possible configuration + call flow scenario. Cheers, -Victor On 16-02-19 02:24 PM, Tomasz Ostrowski wrote: > Hello, > while testing FreeSWITCH 1.4.26 I stumbled upon old and seemingly ignored > problem when FreeSWITCH replies with SDP containing: > > m=image 0 udptl t38 > m=image 0 udptl t38 > > when receiving re-INVITE with disabled audio media and enabled image media > (switching from audio to image). As far as I know this is correct way to > change media, from RFC 3264: > > 8.1 Adding a Media Stream > > New media streams are created by new additional media descriptions > below the existing ones, or by reusing the "slot" used by an old > media stream which had been disabled by setting its port to zero. > > Reusing its slot means that the new media description replaces the > old one, but retains its positioning relative to other media > descriptions in the SDP. New media descriptions MUST appear below > any existing media sections. The rules for formatting these media > descriptions are identical to those described in Section 5. > > When the answerer receives an SDP with more media descriptions than > the previous SDP from the offerer, or it receives an SDP with a media > stream in a slot where the port was previously zero, the answerer > knows that new media streams are being added. These can be rejected > or accepted by placing an appropriately structured media description > in the answer. The procedures for constructing the new media > description in the answer are described in Section 6. > > This problem is mentioned in: > https://freeswitch.org/jira/browse/FS-7037 > https://freeswitch.org/jira/browse/FS-6212 > > With reversed transmission direction (when FreeSWITCH receives FAX and > re-INVITES) re-INVITE contains only image media, but this is easier to > accept (https://freeswitch.org/jira/browse/FS-6954 - correcting it caused > interoperability problems between FS versions), while not accepting > correct SDP by FreeSWITCH is really painful as it requires implementing > RFC non-compliant negotiation and probably adding special switch in > configuration to be interoperable. > > Is this issue fixed in FreeSWITCH 1.6 (I cannot find any further > references in jira)? > Could you give me any suggestions where to look in FreeSWITCH source code > (10k LOC sofia.c seems pretty complex)? > From schoch+freeswitch.org at xwin32.com Wed Feb 24 00:28:01 2016 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 23 Feb 2016 13:28:01 -0800 Subject: [Freeswitch-users] Multiple Choice Caller-ID Message-ID: We use the Polycom SoundPoint IP phones in the office, which work great. One user has 3 different numbers to reach that phone (one is the main office number with an extension, the other 2 ring direct). These phones have 2 line buttons. I would like this user to be able to choose from 3 different Caller-ID numbers when placing a call. I already do this with 2 numbers by using the 2 different line buttons, but I'm trying to figure out a way to choose a 3rd number when placing a call. Any ideas will be appreciated. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/2fc6eaff/attachment.html From brian at freeswitch.org Wed Feb 24 00:49:33 2016 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Feb 2016 15:49:33 -0600 Subject: [Freeswitch-users] Multiple Choice Caller-ID In-Reply-To: References: Message-ID: prefix dialing. On Tue, Feb 23, 2016 at 3:28 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > We use the Polycom SoundPoint IP phones in the office, which work great. > One user has 3 different numbers to reach that phone (one is the main > office number with an extension, the other 2 ring direct). > These phones have 2 line buttons. I would like this user to be able to > choose from 3 different Caller-ID numbers when placing a call. I already do > this with 2 numbers by using the 2 different line buttons, but I'm trying > to figure out a way to choose a 3rd number when placing a call. > Any ideas will be appreciated. > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/ea8588da/attachment.html From ssinyagin at gmail.com Wed Feb 24 02:04:50 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 24 Feb 2016 00:04:50 +0100 Subject: [Freeswitch-users] Multiple Choice Caller-ID In-Reply-To: References: Message-ID: or an IVR that asks you which caller ID to use. Also it may remember your previous choice or use some kind of a rule in your address book... I've done this with a Perl script that looks up in Google Contacts, and the contacts are organized into groups. On Tue, Feb 23, 2016 at 10:49 PM, Brian West wrote: > prefix dialing. > > On Tue, Feb 23, 2016 at 3:28 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> We use the Polycom SoundPoint IP phones in the office, which work great. >> One user has 3 different numbers to reach that phone (one is the main >> office number with an extension, the other 2 ring direct). >> These phones have 2 line buttons. I would like this user to be able to >> choose from 3 different Caller-ID numbers when placing a call. I already do >> this with 2 numbers by using the 2 different line buttons, but I'm trying >> to figure out a way to choose a 3rd number when placing a call. >> Any ideas will be appreciated. >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160224/799ea9f9/attachment.html From msc at freeswitch.org Wed Feb 24 03:44:21 2016 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Feb 2016 16:44:21 -0800 Subject: [Freeswitch-users] Multiple Choice Caller-ID In-Reply-To: References: Message-ID: On Tue, Feb 23, 2016 at 3:04 PM, Stanislav Sinyagin wrote: > or an IVR that asks you which caller ID to use. Also it may remember your > previous choice or use some kind of a rule in your address book... > > I've done this with a Perl script that looks up in Google Contacts, and > the contacts are organized into groups. > > > > On Tue, Feb 23, 2016 at 10:49 PM, Brian West wrote: > >> prefix dialing. >> > +1 Once the user gets accustomed to dialing 7+phone number for caller ID 123 and 8+phone number for caller ID 456 then they'll never go back. (Use whatever prefixes are the best fit for your existing dialing plan.) -MSC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/401797e5/attachment.html From msc at freeswitch.org Wed Feb 24 04:27:12 2016 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Feb 2016 17:27:12 -0800 Subject: [Freeswitch-users] Unable to detect beep with mod_avmd In-Reply-To: References: Message-ID: It definitely works, I just don't think it works on a one-legged call. From fs_cli try this: originate loopback/1001 &playback(silence_stream://60000) You'll have two uuids. Try the avmd app on each one. It will detect it on the "outbound" leg but not the "inbound" leg. -MSC On Tue, Feb 23, 2016 at 5:34 AM, Aqs Younas wrote: > Hi, I am using avmd to detect beep over a channel. Beep is generated by > below script. > > > > data="tone_stream://L=5;%(500,6850,850)"/> > > > > > and starts avmd on fs_cli as. > > freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start > > > I could proper hear beeps over the phone but does not see avmd detecting > this. Does not see anything like "beep detected" in logs. > > I am made several attempts but unable to detect beep even a single time. > > Is there something wrong that i am doing? > > Thanks in advance. > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160223/8d95a604/attachment-0001.html From naveen32india at gmail.com Wed Feb 24 14:40:40 2016 From: naveen32india at gmail.com (Naveen Tamanam) Date: Wed, 24 Feb 2016 17:10:40 +0530 Subject: [Freeswitch-users] Do we have option invalid-sound-long in freeswitch IVR menu ? Message-ID: I would like to play different invalid sound for the first failure which would be long sound and some other invalid sound for next retries. Do we have any kind of option like greet-long and greet-short to play different invalid sounds, like invalid-long invalid-short ? -- Thanks & Regards, Naveen Tamanam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/30566c67/attachment.html From veerabhadrarao.kankatala at panamaxil.com Wed Feb 24 14:55:00 2016 From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala) Date: Wed, 24 Feb 2016 06:55:00 -0500 (EST) Subject: [Freeswitch-users] libyuv problem Message-ID: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com> hello, I am getting following error while configrung freeswitch 1.6.5 ./configure (success) but when i run make command i am facing following error making all mod_fsv make[4]: Entering directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. make[4]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' make[3]: *** [mod_fsv-all] Error 1 make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' make: *** [all] Error 2 but i already installed libyuv packages yum install libyuv-devel Loaded plugins: langpacks, presto, refresh-packagekit Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed and latest version Nothing to do i would like to know which version of libyuv, libyuv-devel is require for freeswitch 1.6.5V I am using Fedora 18 64Bit, Please help me to come out from this problem Thanks in advance -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala at panamaxil.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/6dcb408e/attachment.html From bhadrarao.kankatala at gmail.com Wed Feb 24 14:47:49 2016 From: bhadrarao.kankatala at gmail.com (Kankatala Bhadra Rao) Date: Wed, 24 Feb 2016 17:17:49 +0530 Subject: [Freeswitch-users] Lib yuv problem Message-ID: hello, I am getting following error while configrung freeswitch 1.6.5 ./configure (success) but when i run make command i am facing following error making all mod_fsv make[4]: Entering directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. make[4]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' make[3]: *** [mod_fsv-all] Error 1 make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' make: *** [all] Error 2 but i already installed libyuv packages yum install libyuv-devel Loaded plugins: langpacks, presto, refresh-packagekit Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed and latest version Nothing to do i would like to know which version oflibyuv, libyuv-devel is require for freeswitch 1.6.5V I am using Fedora 18 64Bit, Please help me to come out from this problem Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/7ab90832/attachment.html From mike at jerris.com Wed Feb 24 16:19:30 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Feb 2016 08:19:30 -0500 Subject: [Freeswitch-users] Do we have option invalid-sound-long in freeswitch IVR menu ? In-Reply-To: References: Message-ID: That option doesn't exist currently. Take a look at the code, it shouldn't be really hard to add one as long as not specifying it maintains current behavior. Give a try at a pull request to add the feature. On Wednesday, February 24, 2016, Naveen Tamanam wrote: > I would like to play different invalid sound for the first failure which > would be long sound and some other invalid sound for next retries. > > Do we have any kind of option like greet-long and greet-short to play > different invalid sounds, like invalid-long invalid-short ? > > -- > Thanks & Regards, > Naveen Tamanam > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/c2e17978/attachment.html From mike at jerris.com Wed Feb 24 16:22:32 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Feb 2016 08:22:32 -0500 Subject: [Freeswitch-users] libyuv problem In-Reply-To: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com> References: <1700116939.11698256.1456314900112.JavaMail.zimbra@panamaxil.com> Message-ID: It will only work if you use our libyuv package or tarballs. Fedoras packages are broken, missing the pkg-config files. Note we are in process of moving this library in tree static to not use system libs along with vpx due to needs to have much newer versions changing at a much higher rate than system packages. This change will come very soon. On Wednesday, February 24, 2016, Veerabhadrarao Kankatala < veerabhadrarao.kankatala at panamaxil.com> wrote: > hello, > > I am getting following error while configrung freeswitch 1.6.5. > > ./configure (success) > > but when i run make command i am facing following error > > making all mod_fsv > make[4]: Entering directory > `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' > Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. > make[4]: Leaving directory > `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' > make[3]: *** [mod_fsv-all] Error 1 > make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' > make: *** [all] Error 2 > > but i already installed libyuv packages > > yum install libyuv-devel > Loaded plugins: langpacks, presto, refresh-packagekit > Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed > and latest version > Nothing to do > > > i would like to know which version of libyuv, libyuv-devel is require > for freeswitch 1.6.5V > > I am using Fedora 18 64Bit, > > Please help me to come out from this problem > > Thanks in advance > > -- > Thanks & Regards > *Veerabhadrarao Kankatala* > Software Developer (C & Unix) > *PANAMAX INFOTECH LIMITED* > *Mobile:* *+91-8401231249* > *Messenger Id:* Skype: veerabhadrarao.kankatala > *E-mail: **veerabhadrarao.kankatala at panamaxil.com > * > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/26557879/attachment-0001.html From mylists at polite.se Wed Feb 24 18:23:50 2016 From: mylists at polite.se (Oivvio Polite) Date: Wed, 24 Feb 2016 16:23:50 +0100 Subject: [Freeswitch-users] Joining several FS servers in different regions Message-ID: <20160224152350.GA30297@blomma.liberationtech.net> I'm working on a Saas service that will connect WebRTC (with SIP or Verto for signaling) clients to one another and to PSTN. I'm aware of other such services, but this is tailored to a specific niche. I also want WebRTC media to flow through FS, rather than P2P, so that I can record it. The primary markets are Europe and North America. To keep latency low I need to have servers on both continents. Later on I might want to add more regions. I'm not too worried about high availability or high loads. The clients can keep track of which FS is closest to them and register with that. But this still leaves me with a couple of questions. 1. Maintaining a single user database ===================================== What's the simplest way of maintaining one joint user database? (User directory in FS parlance). The first thing that comes to mind is to generate the xml dynamically from a central database, and force as `reloadxml` on all servers every time there's an update. The communication between central database and FS servers could be via RabbitMQ. mod_xml_curl could also be a part of this I guess. How does that sound? Is there an other simpler way of achiving my goal that I'm not seeing? 2. Keeping track of where a user i currently registered ======================================================= Let's say I have a client Alice who's registered with a FS server in North America and a client Bob who has registered wit a FS server in Europe. Now Alice tries to call Bob. How does the North America server know that Bob is currently registered with the Europe server and that the call should be routed through there? For this second problem I don't even have a tentative solution so I'm curious to hear any ideas. 3. SIP Trunks and geography =========================== If Alice is in North America and wants to call a PSTN endpoint in North America I should obviously be using a SIP trunk in North America, since going over the Atlantic twice will add a lot of latency. But what about when Alice want to call a PSTN endpoint in Europe? Is it better to use a SIP endpoint in Europe or in North America? Thanks in advance Oivvio From ssinyagin at gmail.com Wed Feb 24 18:49:01 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 24 Feb 2016 16:49:01 +0100 Subject: [Freeswitch-users] Joining several FS servers in different regions In-Reply-To: <20160224152350.GA30297@blomma.liberationtech.net> References: <20160224152350.GA30297@blomma.liberationtech.net> Message-ID: On Wed, Feb 24, 2016 at 4:23 PM, Oivvio Polite wrote: > > I'm working on a Saas service that will connect WebRTC (with SIP or Verto for > signaling) clients to one another and to PSTN. I'm aware of other such > services, but this is tailored to a specific niche. I also want WebRTC > media to flow through FS, rather than P2P, so that I can record it. > > The primary markets are Europe and North America. To keep latency low I need > to have servers on both continents. Later on I might want to add more > regions. I'm not too worried about high availability or high loads. > > The clients can keep track of which FS is closest to them and register > with that. But this still leaves me with a couple of questions. > > > > 1. Maintaining a single user database > ===================================== > > What's the simplest way of maintaining one joint user database? (User > directory in FS parlance). > > The first thing that comes to mind is to generate the xml dynamically > from a central database, and force as `reloadxml` on all servers every > time there's an update. The communication between central database and > FS servers could be via RabbitMQ. mod_xml_curl could also be a part of > this I guess. > > How does that sound? Is there an other simpler way of achiving my goal > that I'm not seeing? > You can use mod_xml_curl for this. Every time FS receives a REGISTER message from the user, it would retrieve a piece of XML from your application HTTP service. As these requests need to be answered quite quickly, you would rather install the HTTP server in the same location as FreeSWITCH. Also keep in mind that every call to "user_data" function would trigger an extra HTTP request for the user's XML. The function is quite convenient if you want to get some user-specific parameters from the directory. Then it's up to you how you synchronize the back-end databases. Probably something like https://ipfs.io/ would help. > > > 2. Keeping track of where a user i currently registered > ======================================================= > > Let's say I have a client Alice who's registered with a FS server in > North America and a client Bob who has registered wit a FS server in Europe. > > Now Alice tries to call Bob. How does the North America server know that > Bob is currently registered with the Europe server and that the call > should be routed through there? > > For this second problem I don't even have a tentative solution so I'm > curious to hear any ideas. Each successful registration generates an event (if I'm not mistaken), and an entry in FreeSWITCH's own database. So, you should be able to retrieve the IP address of the FreeSWITCH server where the user got registered. Alternatively, you may have one Kamailio cluster in one location of the world, acting as a SIP registrar. A 250ms round-trip delay is not so critical for SIP traffic. Alternatively, you can have one FreeSWITCH cluster acting as a SIP registrar and call routing engine in media pass-through mode. Then few other FreeSWITCH servers would work as media gateways in different regions. > 3. SIP Trunks and geography > =========================== > > If Alice is in North America and wants to call a PSTN endpoint in North > America I should obviously be using a SIP trunk in North America, since > going over the Atlantic twice will add a lot of latency. > > But what about when Alice want to call a PSTN endpoint in Europe? Is it > better to use a SIP endpoint in Europe or in North America? > This needs testing. I would expect that global VoIP carriers like Voxbone have better control on their bandwidth and transport networks. From dominique.jeannerod at interact-iv.com Wed Feb 24 19:38:28 2016 From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod) Date: Wed, 24 Feb 2016 17:38:28 +0100 Subject: [Freeswitch-users] mod_distributor / gwlist down Message-ID: Hello, i'm currently working with mod_distributor, to load balance calls to gateways. The load balancing works well, and it's rather simple to configure, no problem. I'm trying to automatically exclude down gateways from the load balancing, which seems quite simple, too, when reading the mod_distributor doc... but i can't make it work. I'm using freeswitch 1.6.6. I configured a profile internal_trk_1, and 2 gateways (svc-isr-fr-301, and non_existent_gw), with ping option. One of the gateways, called (non_existent_gw) is down, and detected down by sofia When I check the status of the gateways, the result is ok : sofia profile internal_trk_1 gwlist down non_existent_gw When I check with distributor command in the cli, it's also ok : expand distributor I_MUT_SIPGTW ${sofia profile internal_trk_1 gwlist down} svc-isr-fr-301 But I can't make it work inside the dialplan. I tested using the 2 syntaxes listed in https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor : The dead gateway still gets used by the load balancer. What I am doing wrong ? Best regards Dominique Jeannerod -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/ad0f5ff5/attachment.html From mike at jerris.com Thu Feb 25 02:49:31 2016 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Feb 2016 18:49:31 -0500 Subject: [Freeswitch-users] major build change in master. Message-ID: please note this important change that went into master today. As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries. These are now built static into the freeswitch core. Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it. I'll have more details coming, but let me know if you have any questions. Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week. Mike From Shawn.Wheeler at interlockconcepts.com Tue Feb 23 22:24:16 2016 From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler) Date: Tue, 23 Feb 2016 19:24:16 +0000 Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows? Message-ID: I know there is a windows installer for FreeSWITCH. Does anyone know if this will work with Win 7 32bit? It is what I have a physical Test box Thank you in advance Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160223/cea2ea65/attachment-0001.html From afarooqa at gmail.com Wed Feb 24 01:51:37 2016 From: afarooqa at gmail.com (Faruq Ahmad) Date: Wed, 24 Feb 2016 03:51:37 +0500 Subject: [Freeswitch-users] mod_sms chatplan default behaviour Message-ID: Hi, working on mod_sms, using the demo chatplan called a python script, the script generates a reply for the sms and sends it using chat_execute reply application. however once the script has completed the sms is forwarded to the extensions in "to" field as well. I dont see any application in chatplan that is forwarding the message to the extension. can this feature be controlled? for example if the script called in chatplan has sent a reply for the sms, it is not longer sent to the destination extension? also When sending a message using ESL, logs only show "SMS Delivery assumed successful due to being sent in non-blocking manner" if the destination extension is available. if the destination extensions is not available I get "Nobody to send to" as well as the above message. how can I get any more details for these events. script that is used to send sms has f = con.sendEvent(event) print f.serialize() always shows the following no matter if the message is sent ahead or not. Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK%20accepted -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/748fa3c6/attachment-0001.html From freeswitch at digitaldescent.net Thu Feb 25 06:00:06 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Wed, 24 Feb 2016 19:00:06 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Message-ID: Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@** :46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/e554436b/attachment.html From krice at freeswitch.org Thu Feb 25 06:17:28 2016 From: krice at freeswitch.org (Ken Rice) Date: Wed, 24 Feb 2016 21:17:28 -0600 Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows? In-Reply-To: References: Message-ID: <32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org> There is a newish 64bit installer on files.freeswitch.org but I doubt theres an recent 32 bit one From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler Sent: Tuesday, February 23, 2016 1:24 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows? I know there is a windows installer for FreeSWITCH. Does anyone know if this will work with Win 7 32bit? It is what I have a physical Test box Thank you in advance Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/851fc87f/attachment.html From max at nysolutions.com Thu Feb 25 07:15:36 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 25 Feb 2016 04:15:36 +0000 Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows? In-Reply-To: <32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org> References: <32ae01d16f7b$0f52f310$2df8d930$@freeswitch.org> Message-ID: Win32 installer over a year old http://files.freeswitch.org/windows_installer/installer/x86/freeswitch.msi Win64 installer just over 1 month old http://files.freeswitch.org/windows_installer/installer/x64/freeswitch.msi 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 24, 2016 10:17 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] NUBE to FreeSWITCH - Windows? There is a newish 64bit installer on files.freeswitch.org but I doubt theres an recent 32 bit one From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler Sent: Tuesday, February 23, 2016 1:24 PM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] NUBE to FreeSWITCH - Windows? I know there is a windows installer for FreeSWITCH. Does anyone know if this will work with Win 7 32bit? It is what I have a physical Test box Thank you in advance Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/59cacc51/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/20160225/59cacc51/attachment-0001.jpg From max at nysolutions.com Thu Feb 25 07:31:37 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 25 Feb 2016 04:31:37 +0000 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: Message-ID: <23b1626b3814448b90842a279dc30be5@nysolutions.com> It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on 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 Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@:46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/3db3fae0/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/20160225/3db3fae0/attachment.jpg From Alexander.Haugg at c4b.de Thu Feb 25 09:58:49 2016 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 25 Feb 2016 06:58:49 +0000 Subject: [Freeswitch-users] Web RTC -> No Candidate found In-Reply-To: References: Message-ID: This works, thanks Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West Gesendet: Donnerstag, 18. Februar 2016 15:31 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Web RTC -> No Candidate found Try apply-candidate-acl On Thu, Feb 18, 2016 at 2:29 AM, Alexander Haugg > wrote: Hi, I have the problem, that the freeswitch don?t accept the network internal candidate. The local IP address of the fs is 192.168.241.5 The local IP address of the webrtc Client is 192.168.241.2 In the acl.conf.xml is set: In the sofia profile is set: The following SDP is send by the client: v=0 o=- 222075295642546336 815544600 IN IP4 127.0.0.1 s=IceLink t=0 0 m=audio 30450 RTP/SAVPF 0 8 c=IN IP4 192.168.62.1 a=rtcp:30450 IN IP4 192.168.62.1 a=ice-ufrag:532479e3 a=ice-pwd:4b066f2e64633d218085fc39c49a443c a=sendrecv a=rtcp-mux a=fingerprint:sha-256 FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36 a=setup:actpass a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ssrc:967296980 cname:c2088983 a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 192.168.241.2 44376 typ host a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 192.168.241.2 44376 typ host The output on the fs_cli: 2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 Original read codec set to PCMU:0 2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate ACL defined, Defaulting to wan.auto 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtcp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 no suitable candidates found. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup sofia/H3KSip_B2Bua/170 at 192.168.241.2:5058 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION Thanks a lot! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160225/b7a4496f/attachment-0001.html From findmeinwland at gmail.com Thu Feb 25 11:19:44 2016 From: findmeinwland at gmail.com (Artur Mega) Date: Thu, 25 Feb 2016 13:19:44 +0500 Subject: [Freeswitch-users] serving configuration with mod_lua, and sofia.conf Message-ID: Hello all, I try to serve configuration via lua like it's described at https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+with+Lua In configuration.lua script I set XML_STRING to valid xml, taken from autoload_configs. Similar to this method, i did the same with mod_xml_curl, and all works fine. But with mod_lua sofia profiles not loading. I think, the problem is in variables. Let me explain. When I turn off mod_lua and then make `reload mod_sofia` in fs_cli i see something like ... 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port [5060] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs [OPUS,G722,PCMU,PCMA,VP8] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs [OPUS,G722,PCMU,PCMA,VP8] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [xxx.xx.x.x] .... But when i turn on mod_lua and relod mod_sofia, in logs i see, that variables not calculated! 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port [$${internal_sip_port}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs [$${global_codec_prefs}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs [$${global_codec_prefs}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [$${local_ip_v4}] If i run in fs_cli `eval $${local_ip_v4}`, it prints calculated IP. Maybe I missed something? Thanks -- Arthur ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/040e69a5/attachment.html From mylists at polite.se Thu Feb 25 13:12:13 2016 From: mylists at polite.se (Oivvio Polite) Date: Thu, 25 Feb 2016 11:12:13 +0100 Subject: [Freeswitch-users] Joining several FS servers in different regions In-Reply-To: References: <20160224152350.GA30297@blomma.liberationtech.net> Message-ID: <20160225101213.GB30297@blomma.liberationtech.net> On ons, feb 24, 2016 at 04:49:01 +0100, Stanislav Sinyagin wrote: > You can use mod_xml_curl for this. Every time FS receives a REGISTER > message from the user, it would retrieve a piece of XML from your > application HTTP service. As these requests need to be answered quite > quickly, you would rather install the HTTP server in the same location > as FreeSWITCH. Also keep in mind that every call to "user_data" > function would trigger an extra HTTP request for the user's XML. The > function is quite convenient if you want to get some user-specific > parameters from the directory. > > Then it's up to you how you synchronize the back-end databases. > Probably something like https://ipfs.io/ would help. > > > > Each successful registration generates an event (if I'm not mistaken), > and an entry in FreeSWITCH's own database. So, you should be able to > retrieve the IP address of the FreeSWITCH server where the user got > registered. > > Alternatively, you may have one Kamailio cluster in one location of > the world, acting as a SIP registrar. A 250ms round-trip delay is not > so critical for SIP traffic. > > Alternatively, you can have one FreeSWITCH cluster acting as a SIP > registrar and call routing engine in media pass-through mode. Then few > other FreeSWITCH servers would work as media gateways in different > regions. > > > > This needs testing. I would expect that global VoIP carriers like > Voxbone have better control on their bandwidth and transport networks. > Thanks Stanislav for your suggestions. I think the idea with a Kamailio cluster (or in my case a single box) might be a good fit for my scenario. I'll go read up on that. regards Oivvio From gregor at infomedia.si Thu Feb 25 14:05:45 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 25 Feb 2016 12:05:45 +0100 Subject: [Freeswitch-users] mod_amqp for Windows Message-ID: Hi! Does anybody knows if module mod_amqp is available for windows if I build by myself? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/7d4b40b9/attachment.html From gb at cm.nl Thu Feb 25 14:20:54 2016 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 25 Feb 2016 11:20:54 +0000 Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd Message-ID: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local> Hello, Could someone explain to me the difference between the modules mod_avmd and mod_com_amd, except for the price different? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/91154cda/attachment.html From italo at freeswitch.org Thu Feb 25 14:49:57 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 25 Feb 2016 08:49:57 -0300 Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd In-Reply-To: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local> References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local> Message-ID: mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can analyze patterns of silence and voice and determine if there's an human or a machine in the remote side. On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian wrote: > Hello, > > > > Could someone explain to me the difference between the modules mod_avmd > and mod_com_amd, except for the price different? > > > > Regards, > > > > Grant > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/ef8cfbd1/attachment-0001.html From italo at freeswitch.org Thu Feb 25 14:55:02 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 25 Feb 2016 08:55:02 -0300 Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: What version are you using? Try setting media_mix_inbound_outbound_codecs=true On Tue, Feb 23, 2016 at 12:23 AM, Nandy Dagondon wrote: > Hello guys, > > I installed mod_gsmopen. When testing incoming call, it hangs up with this > error message INCOMPATIBLE_DESTINATION. I understand this is a codec > negotiation issue. > > Looking at the CDR - the gsmopen presented L16 8000 in the SDP. By > default, FS does early negotiation. > > Snippet of my internal sip_profile: > > > > > > > > > Take note: there is no disable-transcoding parameter which I understand FS > is transcodind. > > In vars.xml: > > > > > I tried to add L16 at the end of global_codec_prefs which is unnecessary, > isn't it? But it didn't work, too. > > Finally, I added L16 in the softphone, it worked. > > I have installed gsmopen before and it worked (transcoding) with IP > phones. > > Is there anything I may have missed? I appreciate for any leads. > > Thank you, > > /Nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/d87b3a13/attachment.html From gb at cm.nl Thu Feb 25 15:07:49 2016 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 25 Feb 2016 12:07:49 +0000 Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd In-Reply-To: References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local> Message-ID: <0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local> Thanks! Do both modules give you an event or variable value after ?successful? machine detection, be it using event socket or plain XML? Regards, Grant From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi Sent: Thursday, February 25, 2016 12:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Difference between mod_avmd and mod_com_amd mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can analyze patterns of silence and voice and determine if there's an human or a machine in the remote side. On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian > wrote: Hello, Could someone explain to me the difference between the modules mod_avmd and mod_com_amd, except for the price different? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160225/550f94e6/attachment.html From italo at freeswitch.org Thu Feb 25 15:11:57 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 25 Feb 2016 09:11:57 -0300 Subject: [Freeswitch-users] Difference between mod_avmd and mod_com_amd In-Reply-To: <0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local> References: <8168d1e51be74b4f864a572ebeca6f25@CM-EX-V01.cm.local> <0604e122c6614e549b8b33094acf4b7a@CM-EX-V01.cm.local> Message-ID: Yes, for both. https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd Read the examples. On Thu, Feb 25, 2016 at 9:07 AM, Grant Bagdasarian wrote: > Thanks! > > > > Do both modules give you an event or variable value after ?successful? > machine detection, be it using event socket or plain XML? > > > > Regards, > > > > Grant > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *?talo Rossi > *Sent:* Thursday, February 25, 2016 12:50 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Difference between mod_avmd and > mod_com_amd > > > > mod_avmd can detect the presence of BEEPS on a call and mod_com_amd can > analyze patterns of silence and voice and determine if there's an human or > a machine in the remote side. > > > > On Thu, Feb 25, 2016 at 8:20 AM, Grant Bagdasarian wrote: > > Hello, > > > > Could someone explain to me the difference between the modules mod_avmd > and mod_com_amd, except for the price different? > > > > Regards, > > > > Grant > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > -- ?talo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/9cc61a38/attachment-0001.html From freeswitch at digitaldescent.net Thu Feb 25 08:33:06 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Wed, 24 Feb 2016 21:33:06 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: <23b1626b3814448b90842a279dc30be5@nysolutions.com> References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip. >Is your endpoint registered? How often is it set to reregister? Yes, and the zoiper default registration expiry is 3600 >Does your router have a sip alg? Are the ports opened in your firewall? > ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip. Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein wrote: > It is probably sending call to your extension at domainname, if your > external ip is your domain name then you will see the external ip. > > Is your endpoint registered? How often is it set to reregister? > > Does your router have a sip alg? Are the ports opened in your firewall? > > Very hard to guess without a log of a call with debug enabled. > > sofia global sip trace on > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Wednesday, February 24, 2016 10:00 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Hello all! > > I'm new to freeswitch, so I'm sure this is just a newbie configuration > error. Sorry if it's been answered a million times, my google searches > always just bring up the standard NAT configuration pages. I've already > followed the confluence page on configuring NAT. > > > > My Setup: > > Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 > instance. > > SIP Provider: Flowroute > > > > NAT: extension and freeswitch are on the same network, both of which are > behind NAT and connecting to flowroute. I configured external sip and rtp > to use stun entries. External profile is using $${external_rtp/sip_ip} for > ext-rtp/sip respectively. > > > > I have one DID configured to go directly to my one extension. My > extension can register just fine. My extension can dial out. When I call > my cell, the call connects and I have 2 way audio. When I dial my DID from > my cell, I can see the call hitting the FS server, but instead of ringing > my extension, it goes straight to my extensions voicemail (which I can just > fine). > > > > When I look the at the console, it appears (sorry if this is wrong, I'm > only a day into free switch) that FS is attempting to route the call to my > extension...@ my external ip? > > > > I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] > switch_channel.c:1101 New Channel sofia/internal/@ > **:46072 > > > > > > Shouldn't that read sofia/internal/@ ? > > > > Sofia Status says: > > > > Name Type > Data State > > > ================================================================================================= > > external-ipv6 profile > sip:mod_sofia@[::1]:5080 RUNNING (0) > > external profile > sip:mod_sofia@:5080 RUNNING (0) > > external:: gateway sip:@ > sip.flowroute.com REGED > > internal-ipv6 profile > sip:mod_sofia@[::1]:5060 RUNNING (0) > > internal profile > sip:mod_sofia@:5060 RUNNING (0) > > > ================================================================================================= > > > > If I'm completely off base here, can anyone recommend where I can start > looking to change/troubleshoot the issue? I feel like it's just me missing > something, I just can't determine what that might be. > > > > Thanks, > > -Brian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160224/ff9e8994/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160224/ff9e8994/attachment-0001.jpg From freeswitch.opencode at spamgourmet.com Thu Feb 25 10:35:57 2016 From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com) Date: Thu, 25 Feb 2016 07:35:57 +0000 Subject: [Freeswitch-users] Getting 503 when starting automatically via systemd Message-ID: I'm using Arch Linux (specifically Arch Linux ARM on a Raspberry Pi Zero). I'm successfully done a make and install for freeswitch 1.6.6 and when I run it manually (freeswitch -nonat), it works fine. Also, when I start it manually via systemd (systemctl start freeswitch), it still works fine. However, when I enable freeswitch.service to start automatically and reboot, the freeswitch log shows 503 errors trying to register out to my gateway, and my internal client (an Obi200) gets a 503 when trying to register in to FreeSWITCH as well; this keeps happening repeatedly (waiting 30 seconds and then failing again). However, if I restart the service (systemctl restart freeswitch), connectivity then starts working. Would the 503 be what's reported up if there's no network connectivity either way? I'm using the Debian systemd *freeswitch.service file (with very minor tweaks for my configuration on Arch Linux). Has anyone encountered this problem before (on Debian, Arch Linux, or elsewhere)? The overall .service file is very similar to sshd.service, which doesn't have connection problems, so my guess is there's something specific to FreeSWITCH going on. If this is Arch Linux-specific and I ask in their forums, is there anything different about how FreeSWITCH handles connectivity (perhaps caching some kind of handle) that would help point things in the right direction? Thanks, David From alessandro.illiano at toctoc.me Thu Feb 25 14:15:03 2016 From: alessandro.illiano at toctoc.me (Alessandro Illiano) Date: Thu, 25 Feb 2016 12:15:03 +0100 Subject: [Freeswitch-users] mod_rtmp rtmps support Message-ID: <56CEE237.6060108@toctoc.me> Hi all, does mod_rtmp support rtmps? Thanks Alessandro From brian at freeswitch.org Thu Feb 25 16:50:37 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Feb 2016 07:50:37 -0600 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow wrote: > Debug console output: https://pastebin.freeswitch.org/24572 > > My fqdn only resolves inside the network, but the external-ip is > configured in FS to use stun, which reports my external ip. > > >Is your endpoint registered? How often is it set to reregister? > > Yes, and the zoiper default registration expiry is 3600 > > > > >Does your router have a sip alg? Are the ports opened in your firewall? > > > ish, no GUI for it, but I did make sure to disable it from the command > line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but > allow bi-directional with existing states. I do get 2 way audio when > calling out, and I also can hear the system forwarding my call in to > voicemail; the voicemail message and prompts. I know asterisk is different > from FS, but my asterisk experience has proven that at least with asterisk, > port forwarding isn't needed. Also, unfortunately, I cannot get a static > ip. > > > Thanks! > > -Brian > > On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: > >> It is probably sending call to your extension at domainname, if your >> external ip is your domain name then you will see the external ip. >> >> Is your endpoint registered? How often is it set to reregister? >> >> Does your router have a sip alg? Are the ports opened in your firewall? >> >> Very hard to guess without a log of a call with debug enabled. >> >> sofia global sip trace on >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com >> * >> >> [image: 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 *Brian Chow >> *Sent:* Wednesday, February 24, 2016 10:00 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to >> internal extension with external IP? >> >> >> >> Hello all! >> >> I'm new to freeswitch, so I'm sure this is just a newbie configuration >> error. Sorry if it's been answered a million times, my google searches >> always just bring up the standard NAT configuration pages. I've already >> followed the confluence page on configuring NAT. >> >> >> >> My Setup: >> >> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 >> instance. >> >> SIP Provider: Flowroute >> >> >> >> NAT: extension and freeswitch are on the same network, both of which are >> behind NAT and connecting to flowroute. I configured external sip and rtp >> to use stun entries. External profile is using $${external_rtp/sip_ip} for >> ext-rtp/sip respectively. >> >> >> >> I have one DID configured to go directly to my one extension. My >> extension can register just fine. My extension can dial out. When I call >> my cell, the call connects and I have 2 way audio. When I dial my DID from >> my cell, I can see the call hitting the FS server, but instead of ringing >> my extension, it goes straight to my extensions voicemail (which I can just >> fine). >> >> >> >> When I look the at the console, it appears (sorry if this is wrong, I'm >> only a day into free switch) that FS is attempting to route the call to my >> extension...@ my external ip? >> >> >> >> I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] >> switch_channel.c:1101 New Channel sofia/internal/@ >> **:46072 >> >> >> >> >> >> Shouldn't that read sofia/internal/@ ? >> >> >> >> Sofia Status says: >> >> >> >> Name Type >> Data State >> >> >> ================================================================================================= >> >> external-ipv6 profile >> sip:mod_sofia@[::1]:5080 RUNNING (0) >> >> external profile >> sip:mod_sofia@:5080 RUNNING (0) >> >> external:: gateway sip:@ >> sip.flowroute.com REGED >> >> internal-ipv6 profile >> sip:mod_sofia@[::1]:5060 RUNNING (0) >> >> internal profile >> sip:mod_sofia@:5060 RUNNING (0) >> >> >> ================================================================================================= >> >> >> >> If I'm completely off base here, can anyone recommend where I can start >> looking to change/troubleshoot the issue? I feel like it's just me missing >> something, I just can't determine what that might be. >> >> >> >> Thanks, >> >> -Brian >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160225/0d03a84e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/0d03a84e/attachment-0001.jpg From freeswitch at digitaldescent.net Thu Feb 25 18:59:48 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Thu, 25 Feb 2016 07:59:48 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" wrote: > Ok, in that case, what do I look at next to see why it goes straight to > voicemail instead of ringing my extension? > On Feb 25, 2016 5:52 AM, "Brian West" wrote: > >> Its just the channel name so don't get tripped up by that, its just named >> with some default data about the session. >> >> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow < >> freeswitch at digitaldescent.net> wrote: >> >>> Debug console output: https://pastebin.freeswitch.org/24572 >>> >>> My fqdn only resolves inside the network, but the external-ip is >>> configured in FS to use stun, which reports my external ip. >>> >>> >Is your endpoint registered? How often is it set to reregister? >>> >>> Yes, and the zoiper default registration expiry is 3600 >>> >>> >>> >>> >Does your router have a sip alg? Are the ports opened in your firewall? >>> >>> > ish, no GUI for it, but I did make sure to disable it from the command >>> line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but >>> allow bi-directional with existing states. I do get 2 way audio when >>> calling out, and I also can hear the system forwarding my call in to >>> voicemail; the voicemail message and prompts. I know asterisk is different >>> from FS, but my asterisk experience has proven that at least with asterisk, >>> port forwarding isn't needed. Also, unfortunately, I cannot get a static >>> ip. >>> >>> >>> Thanks! >>> >>> -Brian >>> >>> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein >>> wrote: >>> >>>> It is probably sending call to your extension at domainname, if your >>>> external ip is your domain name then you will see the external ip. >>>> >>>> Is your endpoint registered? How often is it set to reregister? >>>> >>>> Does your router have a sip alg? Are the ports opened in your firewall? >>>> >>>> Very hard to guess without a log of a call with debug enabled. >>>> >>>> sofia global sip trace on >>>> >>>> >>>> >>>> Thanks, >>>> >>>> >>>> >>>> Moishe Grunstein >>>> >>>> Tornado Computer Systems, Inc. >>>> >>>> 212.400.7650 888.IPPBX.US >>>> *Service Request Email: support at nysolutions.com >>>> * >>>> >>>> [image: 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 *Brian >>>> Chow >>>> *Sent:* Wednesday, February 24, 2016 10:00 PM >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to >>>> internal extension with external IP? >>>> >>>> >>>> >>>> Hello all! >>>> >>>> I'm new to freeswitch, so I'm sure this is just a newbie >>>> configuration error. Sorry if it's been answered a million times, my >>>> google searches always just bring up the standard NAT configuration pages. >>>> I've already followed the confluence page on configuring NAT. >>>> >>>> >>>> >>>> My Setup: >>>> >>>> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 >>>> instance. >>>> >>>> SIP Provider: Flowroute >>>> >>>> >>>> >>>> NAT: extension and freeswitch are on the same network, both of which >>>> are behind NAT and connecting to flowroute. I configured external sip and >>>> rtp to use stun entries. External profile is using $${external_rtp/sip_ip} >>>> for ext-rtp/sip respectively. >>>> >>>> >>>> >>>> I have one DID configured to go directly to my one extension. My >>>> extension can register just fine. My extension can dial out. When I call >>>> my cell, the call connects and I have 2 way audio. When I dial my DID from >>>> my cell, I can see the call hitting the FS server, but instead of ringing >>>> my extension, it goes straight to my extensions voicemail (which I can just >>>> fine). >>>> >>>> >>>> >>>> When I look the at the console, it appears (sorry if this is wrong, I'm >>>> only a day into free switch) that FS is attempting to route the call to my >>>> extension...@ my external ip? >>>> >>>> >>>> >>>> I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] >>>> switch_channel.c:1101 New Channel sofia/internal/@ >>>> **:46072 >>>> >>>> >>>> >>>> >>>> >>>> Shouldn't that read sofia/internal/@ ? >>>> >>>> >>>> >>>> Sofia Status says: >>>> >>>> >>>> >>>> Name Type >>>> Data State >>>> >>>> >>>> ================================================================================================= >>>> >>>> external-ipv6 profile >>>> sip:mod_sofia@[::1]:5080 RUNNING (0) >>>> >>>> external profile >>>> sip:mod_sofia@:5080 RUNNING (0) >>>> >>>> external:: gateway >>>> sip:@sip.flowroute.com REGED >>>> >>>> internal-ipv6 profile >>>> sip:mod_sofia@[::1]:5060 RUNNING (0) >>>> >>>> internal profile >>>> sip:mod_sofia@:5060 RUNNING (0) >>>> >>>> >>>> ================================================================================================= >>>> >>>> >>>> >>>> If I'm completely off base here, can anyone recommend where I can start >>>> looking to change/troubleshoot the issue? I feel like it's just me missing >>>> something, I just can't determine what that might be. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Brian >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20160225/10cebed7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/10cebed7/attachment-0001.jpg From max at nysolutions.com Thu Feb 25 19:12:36 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 25 Feb 2016 16:12:36 +0000 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: Does internal call also go direct to voicemail? Are you sure you don?t have a *99 before the extension number? 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 Brian Chow Sent: Thursday, February 25, 2016 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension? On Feb 25, 2016 5:52 AM, "Brian West" > wrote: Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow > wrote: Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip. >Is your endpoint registered? How often is it set to reregister? Yes, and the zoiper default registration expiry is 3600 >Does your router have a sip alg? Are the ports opened in your firewall? > ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip. Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on 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 Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@:46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160225/17aa0b64/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/20160225/17aa0b64/attachment-0001.jpg From aqsyounas at gmail.com Thu Feb 25 19:37:26 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 25 Feb 2016 21:37:26 +0500 Subject: [Freeswitch-users] Unable to detect beep with mod_avmd In-Reply-To: References: Message-ID: Thank You Much. I was testing it on single leg call. Starting avmd on outbound leg(of two leg call) perfectly detected the beep. This email has been sent from a virus-free computer protected by Avast. www.avast.com <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On 24 February 2016 at 06:27, Michael Collins wrote: > It definitely works, I just don't think it works on a one-legged call. > From fs_cli try this: > > originate loopback/1001 &playback(silence_stream://60000) > > You'll have two uuids. Try the avmd app on each one. It will detect it on > the "outbound" leg but not the "inbound" leg. > > -MSC > > On Tue, Feb 23, 2016 at 5:34 AM, Aqs Younas wrote: > >> Hi, I am using avmd to detect beep over a channel. Beep is generated by >> below script. >> >> >> >> > data="tone_stream://L=5;%(500,6850,850)"/> >> >> >> >> >> and starts avmd on fs_cli as. >> >> freeswitch at debian>avmd 30d99731-4ff3-4f02-ad4b-85a3a8b9d600 start >> >> >> I could proper hear beeps over the phone but does not see avmd detecting >> this. Does not see anything like "beep detected" in logs. >> >> I am made several attempts but unable to detect beep even a single time. >> >> Is there something wrong that i am doing? >> >> Thanks in advance. >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160225/78832b32/attachment.html From joelists at tm.net.uk Thu Feb 25 20:00:08 2016 From: joelists at tm.net.uk (Joseph Waite) Date: Thu, 25 Feb 2016 17:00:08 +0000 Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration Message-ID: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk> Evening all Does anyone know of an integration with sales force? Looking for click to dial and screen pops primarily. With some kind of call recording integration being ideal. Open source preferred, commercial solutions considered! Joe Waite From aqsyounas at gmail.com Thu Feb 25 20:13:48 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 25 Feb 2016 22:13:48 +0500 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep Message-ID: Hi, I have subscribed to CUSTOM avmd::beep and this is what I get. "Event-Subclass": "avmd::beep", "Event-Name": "CUSTOM", "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", "FreeSWITCH-Hostname": "debian", "FreeSWITCH-Switchname": "debian", "FreeSWITCH-IPv4": "192.168.10.59", "FreeSWITCH-IPv6": "::1", "Event-Date-Local": "2016-02-25 10:46:17", "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", "Event-Date-Timestamp": "1456415177487145", "Event-Calling-File": "mod_avmd.c", "Event-Calling-Function": "avmd_process", "Event-Calling-Line-Number": "550", "Event-Sequence": "3174", "Beep-Status": "stop", "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", "call-command": "avmd" Now I need to hangup that call(beep detected), but could not find any event variable connected to the call. How can i kill the call. Any pointer is much appreciated. This email has been sent from a virus-free computer protected by Avast. www.avast.com <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/a7029460/attachment.html From max at nysolutions.com Thu Feb 25 20:26:08 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 25 Feb 2016 17:26:08 +0000 Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration In-Reply-To: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk> References: <90B18E57-F6AF-405F-A65B-A034F2AAE069@tm.net.uk> Message-ID: <8e62b7d38f7e401eb73f10ad61120152@nysolutions.com> There are several solutions that work with any system, they talk to the endpoint. You can look at http://camrivox.com/products/flexor-cti-salesforce/ Not opensource Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph Waite Sent: Thursday, February 25, 2016 12:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch/FusionPBX sales force integration Evening all Does anyone know of an integration with sales force? Looking for click to dial and screen pops primarily. With some kind of call recording integration being ideal. Open source preferred, commercial solutions considered! Joe Waite _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 mario_fs at mgtech.com Thu Feb 25 21:16:15 2016 From: mario_fs at mgtech.com (Mario G) Date: Thu, 25 Feb 2016 10:16:15 -0800 Subject: [Freeswitch-users] major build change in master. In-Reply-To: References: Message-ID: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> This fixed OS X VPX issue, aok now. Questions: I assume this means YUV and VPX can be removed as prerequisites? Will this be one for 1.6 as well? Just asking so I can update the wiki and Applescript installer as needed. Thanks! > On Feb 24, 2016, at 3:49 PM, Michael Jerris wrote: > > please note this important change that went into master today. As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries. These are now built static into the freeswitch core. Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it. I'll have more details coming, but let me know if you have any questions. Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week. > > Mike > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Thu Feb 25 21:26:09 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 25 Feb 2016 13:26:09 -0500 Subject: [Freeswitch-users] Getting 503 when starting automatically via systemd In-Reply-To: References: Message-ID: <01de01d16ff9$ffe124e0$ffa36ea0$@botecomm.com> A 503 error, huh? Weird. Check for permissions problems. My guess is that you are building as "root" so when you first ran it manually all the databases and log files were created with root as owner. But when it boots systemd is starting it as root and then dropping privileges to the freeswitch user as it should. After dropping privileges it can't access its config files or open ports necessary to function. Again, this is just my S.W.A.G. but it has happened enough times in the past that it's a pretty good guess. If it bears out you might want to check the instructions on this deprecated page: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#Fr eeSWITCH1.6Video-StartingFreeSWITCH Also, once you achieve success there are MANY other RPi users out there who would love to see your recipe for success. https://freeswitch.org/confluence/display/FREESWITCH/Raspberry+Pi Hope this helps. Bote > -----Original Message----- > From: freeswitch.opencode at spamgourmet.com > Sent: Thursday, 25 February, 2016 02:36 > Subject: [Freeswitch-users] Getting 503 when starting automatically via systemd > > I'm using Arch Linux (specifically Arch Linux ARM on a Raspberry Pi Zero). I'm > successfully done a make and install for freeswitch 1.6.6 and when I run it > manually (freeswitch -nonat), it works fine. Also, when I start it manually via > systemd (systemctl start freeswitch), it still works fine. > > However, when I enable freeswitch.service to start automatically and > reboot, the freeswitch log shows 503 errors trying to register out to my > gateway, and my internal client (an Obi200) gets a 503 when trying to register > in to FreeSWITCH as well; this keeps happening repeatedly (waiting 30 > seconds and then failing again). However, if I restart the service (systemctl > restart freeswitch), connectivity then starts working. > > Would the 503 be what's reported up if there's no network connectivity > either way? > > I'm using the Debian systemd *freeswitch.service file (with very minor > tweaks for my configuration on Arch Linux). Has anyone encountered this > problem before (on Debian, Arch Linux, or elsewhere)? The overall .service > file is very similar to sshd.service, which doesn't have connection problems, > so my guess is there's something specific to FreeSWITCH going on. If this is > Arch Linux-specific and I ask in their forums, is there anything different about > how FreeSWITCH handles connectivity (perhaps caching some kind of > handle) that would help point things in the right direction? > > Thanks, > > David > From bote_radio at botecomm.com Thu Feb 25 21:39:27 2016 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 25 Feb 2016 13:39:27 -0500 Subject: [Freeswitch-users] major build change in master. In-Reply-To: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> References: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> Message-ID: <01e201d16ffb$db648f10$922dad30$@botecomm.com> Mike, I put a note to this effect on the Debian 8 installation page. I'm glad I caught this as I have been on a customer's site a lot and have not been able to keep with the -users mailing list. Lemme know if this changes so I can update it. Thanks. Bote > -----Original Message----- > From: Mario G > Sent: Thursday, 25 February, 2016 13:16 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] major build change in master. > > This fixed OS X VPX issue, aok now. Questions: > > I assume this means YUV and VPX can be removed as prerequisites? > > Will this be one for 1.6 as well? > > Just asking so I can update the wiki and Applescript installer as needed. > Thanks! > > > > On Feb 24, 2016, at 3:49 PM, Michael Jerris wrote: > > > > please note this important change that went into master today. As of > today, we are no longer using system versions of libyuv and libvpx due to > major conflicts with system versions of these libraries. These are now built > static into the freeswitch core. Also note, mod_vpx no longer exists, it is > automatically loaded as part of the core and you will no longer have > mod_vpx.so or have to manually load it. I'll have more details coming, but let > me know if you have any questions. Please note for anyone doing cross > compiling, this probably needs a bit more work for you, I'll be working on > fixing that this week. > > > > Mike > > From blasterjr at gmail.com Thu Feb 25 21:43:10 2016 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 25 Feb 2016 11:43:10 -0700 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep In-Reply-To: References: Message-ID: uuid_hangup the value of the Unique-ID field possibly? On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas wrote: > Hi, > > I have subscribed to CUSTOM avmd::beep and this is what I get. > > "Event-Subclass": "avmd::beep", > "Event-Name": "CUSTOM", > "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", > "FreeSWITCH-Hostname": "debian", > "FreeSWITCH-Switchname": "debian", > "FreeSWITCH-IPv4": "192.168.10.59", > "FreeSWITCH-IPv6": "::1", > "Event-Date-Local": "2016-02-25 10:46:17", > "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", > "Event-Date-Timestamp": "1456415177487145", > "Event-Calling-File": "mod_avmd.c", > "Event-Calling-Function": "avmd_process", > "Event-Calling-Line-Number": "550", > "Event-Sequence": "3174", > "Beep-Status": "stop", > "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", > "call-command": "avmd" > > Now I need to hangup that call(beep detected), but could not find any > event variable connected to the call. > > How can i kill the call. > > Any pointer is much appreciated. > > This email has been sent from a virus-free computer protected by Avast. > www.avast.com > > <#-1458571440_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/20160225/2cfa69c0/attachment.html From mike at jerris.com Thu Feb 25 21:48:27 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Feb 2016 13:48:27 -0500 Subject: [Freeswitch-users] major build change in master. In-Reply-To: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> References: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> Message-ID: <2C14558E-FBD9-4532-A1FC-2272DF36DB71@jerris.com> This will remove the prereqs for all new versions released after this date, so the newest 1.6 for example will not need those anymore, all previous requirements for previous versions stay the same. > On Feb 25, 2016, at 1:16 PM, Mario G wrote: > > This fixed OS X VPX issue, aok now. Questions: > > I assume this means YUV and VPX can be removed as prerequisites? > > Will this be one for 1.6 as well? > > Just asking so I can update the wiki and Applescript installer as needed. Thanks! > > >> On Feb 24, 2016, at 3:49 PM, Michael Jerris wrote: >> >> please note this important change that went into master today. As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries. These are now built static into the freeswitch core. Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it. I'll have more details coming, but let me know if you have any questions. Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week. >> >> Mike From mike at jerris.com Thu Feb 25 21:49:33 2016 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Feb 2016 13:49:33 -0500 Subject: [Freeswitch-users] major build change in master. In-Reply-To: <01e201d16ffb$db648f10$922dad30$@botecomm.com> References: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> <01e201d16ffb$db648f10$922dad30$@botecomm.com> Message-ID: <8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com> we are working on how exactly this will effect packaging and install instructions, but we will no longer have yuv or vpx pre-reqs and the requirement to rebuild libav against our vpx is also removed. > On Feb 25, 2016, at 1:39 PM, Bote Man wrote: > > Mike, I put a note to this effect on the Debian 8 installation page. > > I'm glad I caught this as I have been on a customer's site a lot and have > not been able to keep with the -users mailing list. Lemme know if this > changes so I can update it. > > Thanks. > > Bote > > >> -----Original Message----- >> From: Mario G >> Sent: Thursday, 25 February, 2016 13:16 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] major build change in master. >> >> This fixed OS X VPX issue, aok now. Questions: >> >> I assume this means YUV and VPX can be removed as prerequisites? >> >> Will this be one for 1.6 as well? >> >> Just asking so I can update the wiki and Applescript installer as needed. >> Thanks! >> >> >>> On Feb 24, 2016, at 3:49 PM, Michael Jerris wrote: >>> >>> please note this important change that went into master today. As of >> today, we are no longer using system versions of libyuv and libvpx due to >> major conflicts with system versions of these libraries. These are now > built >> static into the freeswitch core. Also note, mod_vpx no longer exists, it > is >> automatically loaded as part of the core and you will no longer have >> mod_vpx.so or have to manually load it. I'll have more details coming, > but let >> me know if you have any questions. Please note for anyone doing cross >> compiling, this probably needs a bit more work for you, I'll be working on >> fixing that this week. >>> >>> Mike >>> From aqsyounas at gmail.com Thu Feb 25 23:01:12 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 26 Feb 2016 01:01:12 +0500 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep In-Reply-To: References: Message-ID: Thanks for your Reply. freeswitch at debian> uuid_hangup Unique-ID Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc Tried to kill. freeswitch at debian> uuid_kill Unique-ID -ERR No such channel! Any suggestion? On 25 February 2016 at 23:43, Chris Tunbridge wrote: > uuid_hangup the value of the Unique-ID field possibly? > > On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas wrote: > >> Hi, >> >> I have subscribed to CUSTOM avmd::beep and this is what I get. >> >> "Event-Subclass": "avmd::beep", >> "Event-Name": "CUSTOM", >> "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", >> "FreeSWITCH-Hostname": "debian", >> "FreeSWITCH-Switchname": "debian", >> "FreeSWITCH-IPv4": "192.168.10.59", >> "FreeSWITCH-IPv6": "::1", >> "Event-Date-Local": "2016-02-25 10:46:17", >> "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", >> "Event-Date-Timestamp": "1456415177487145", >> "Event-Calling-File": "mod_avmd.c", >> "Event-Calling-Function": "avmd_process", >> "Event-Calling-Line-Number": "550", >> "Event-Sequence": "3174", >> "Beep-Status": "stop", >> "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", >> "call-command": "avmd" >> >> Now I need to hangup that call(beep detected), but could not find any >> event variable connected to the call. >> >> How can i kill the call. >> >> Any pointer is much appreciated. >> >> This email has been sent from a virus-free computer protected by Avast. >> www.avast.com >> >> <#-2019354057_-1458571440_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/20160226/ccbdbe89/attachment-0001.html From nandy1925 at gmail.com Thu Feb 25 23:58:53 2016 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 26 Feb 2016 04:58:53 +0800 Subject: [Freeswitch-users] gsmopen problem: INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: I installed version 1.7. Somehow I got a tip to use absolute_codec_string in my dialplan. "set"ting failed. Realizing that non-L16 codecs has to be in B-leg, "export"ing absolute_codec_string worked. I'll try your suggestion because the variable is new to me. Thanks! ==================================================== MagicBox.PH - *keeps Filipino families in touch worldwide for free*Lapulapu City, Phils Phone: +63-32-3401807, Mobile: +63-920-6373450 *USA DID# :* (646)547-1225 *Worldwide:* [**011*][*63-32-3401807#*] via any 200+ Access Numbers (37 Countries) On Thu, Feb 25, 2016 at 7:55 PM, ?talo Rossi wrote: > What version are you using? Try setting > media_mix_inbound_outbound_codecs=true > > On Tue, Feb 23, 2016 at 12:23 AM, Nandy Dagondon > wrote: > >> Hello guys, >> >> I installed mod_gsmopen. When testing incoming call, it hangs up with >> this error message INCOMPATIBLE_DESTINATION. I understand this is a codec >> negotiation issue. >> >> Looking at the CDR - the gsmopen presented L16 8000 in the SDP. By >> default, FS does early negotiation. >> >> Snippet of my internal sip_profile: >> >> >> >> >> >> >> >> >> Take note: there is no disable-transcoding parameter which I understand >> FS is transcodind. >> >> In vars.xml: >> >> >> >> >> I tried to add L16 at the end of global_codec_prefs which is unnecessary, >> isn't it? But it didn't work, too. >> >> Finally, I added L16 in the softphone, it worked. >> >> I have installed gsmopen before and it worked (transcoding) with IP >> phones. >> >> Is there anything I may have missed? I appreciate for any leads. >> >> Thank you, >> >> /Nandy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160226/3d52ff2f/attachment.html From blasterjr at gmail.com Fri Feb 26 00:10:54 2016 From: blasterjr at gmail.com (Chris Tunbridge) Date: Thu, 25 Feb 2016 14:10:54 -0700 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep In-Reply-To: References: Message-ID: My guess is the UUID of that call is not the same as the Unique-ID, i don't think it'd be Core-UUID either. I haven't personally played with mod_amvd or the events from it. On Thu, Feb 25, 2016 at 1:01 PM, Aqs Younas wrote: > Thanks for your Reply. > > freeswitch at debian> uuid_hangup Unique-ID > Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc > > Tried to kill. > > freeswitch at debian> uuid_kill Unique-ID > > -ERR No such channel! > > Any suggestion? > > On 25 February 2016 at 23:43, Chris Tunbridge wrote: > >> uuid_hangup the value of the Unique-ID field possibly? >> >> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas wrote: >> >>> Hi, >>> >>> I have subscribed to CUSTOM avmd::beep and this is what I get. >>> >>> "Event-Subclass": "avmd::beep", >>> "Event-Name": "CUSTOM", >>> "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", >>> "FreeSWITCH-Hostname": "debian", >>> "FreeSWITCH-Switchname": "debian", >>> "FreeSWITCH-IPv4": "192.168.10.59", >>> "FreeSWITCH-IPv6": "::1", >>> "Event-Date-Local": "2016-02-25 10:46:17", >>> "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", >>> "Event-Date-Timestamp": "1456415177487145", >>> "Event-Calling-File": "mod_avmd.c", >>> "Event-Calling-Function": "avmd_process", >>> "Event-Calling-Line-Number": "550", >>> "Event-Sequence": "3174", >>> "Beep-Status": "stop", >>> "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", >>> "call-command": "avmd" >>> >>> Now I need to hangup that call(beep detected), but could not find any >>> event variable connected to the call. >>> >>> How can i kill the call. >>> >>> Any pointer is much appreciated. >>> >>> This email has been sent from a virus-free computer protected by Avast. >>> www.avast.com >>> >>> <#-1363453803_-2019354057_-1458571440_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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/7d314d0b/attachment-0001.html From aqsyounas at gmail.com Fri Feb 26 00:15:05 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 26 Feb 2016 02:15:05 +0500 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep In-Reply-To: References: Message-ID: My fault. Got it working with uuid_kill Unique-ID Thanks. On 26 February 2016 at 01:01, Aqs Younas wrote: > Thanks for your Reply. > > freeswitch at debian> uuid_hangup Unique-ID > Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc > > Tried to kill. > > freeswitch at debian> uuid_kill Unique-ID > > -ERR No such channel! > > Any suggestion? > > On 25 February 2016 at 23:43, Chris Tunbridge wrote: > >> uuid_hangup the value of the Unique-ID field possibly? >> >> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas wrote: >> >>> Hi, >>> >>> I have subscribed to CUSTOM avmd::beep and this is what I get. >>> >>> "Event-Subclass": "avmd::beep", >>> "Event-Name": "CUSTOM", >>> "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", >>> "FreeSWITCH-Hostname": "debian", >>> "FreeSWITCH-Switchname": "debian", >>> "FreeSWITCH-IPv4": "192.168.10.59", >>> "FreeSWITCH-IPv6": "::1", >>> "Event-Date-Local": "2016-02-25 10:46:17", >>> "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", >>> "Event-Date-Timestamp": "1456415177487145", >>> "Event-Calling-File": "mod_avmd.c", >>> "Event-Calling-Function": "avmd_process", >>> "Event-Calling-Line-Number": "550", >>> "Event-Sequence": "3174", >>> "Beep-Status": "stop", >>> "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", >>> "call-command": "avmd" >>> >>> Now I need to hangup that call(beep detected), but could not find any >>> event variable connected to the call. >>> >>> How can i kill the call. >>> >>> Any pointer is much appreciated. >>> >>> This email has been sent from a virus-free computer protected by Avast. >>> www.avast.com >>> >>> <#996209302_-2019354057_-1458571440_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/20160226/92f23ff6/attachment.html From freeswitch.opencode at spamgourmet.com Thu Feb 25 22:01:41 2016 From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com) Date: Thu, 25 Feb 2016 19:01:41 +0000 Subject: [Freeswitch-users] Getting 503 when starting automatically Message-ID: On Thu, 2016-02-25 at 21:26:09 +0300, Bote wrote: > A 503 error, huh? Weird. Check for permissions problems. > My guess is that you are building as "root" so when you first ran it > manually all the databases and log files were created with root as owner. > But when it boots systemd is starting it as root and then dropping > privileges to the freeswitch user as it should. After dropping privileges it > can't access its config files or open ports necessary to function. Thanks for the suggestion. I compiled as a normal user using Arch Linux's makepkg tool and installed using its pacman tool, so I don't think that's the problem. I'm also letting system start freeswitch.service with its default user setting (as root). The weirdest thing is that it does start and connect fine using systemd when make a request after boot to have systemd start it (systemctl start freeswitch or systemctl restart freeswitch). In both cases, systemd is controlling process identity, so I don't think it's a permissions issue. My best guess is there's some kind of race condition at boot and if FreeSWITCH starts too early, it gets a cached but invalid handle of some sort that doesn't allow communicating in or out. This is just a guess though. Thanks, David From lists at kavun.ch Fri Feb 26 02:13:42 2016 From: lists at kavun.ch (Emrah) Date: Fri, 26 Feb 2016 00:13:42 +0100 Subject: [Freeswitch-users] SRTP breaks my TLS session Message-ID: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> Hello list, I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size. Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS. Incoming calls work reliably with or without SRTP. How do you suggest debugging this? I tried setting up a fresh instance of FS but the issue persists. Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device. Any suggestion is welcome. Have you experienced this? I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95. E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/f16199a2/attachment.html From sharigosrl at gmail.com Fri Feb 26 01:24:41 2016 From: sharigosrl at gmail.com (sharigo roma) Date: Thu, 25 Feb 2016 23:24:41 +0100 Subject: [Freeswitch-users] discriminate between session answered by users and voice mail automatic messages Message-ID: Dear freeswitchers, is there a way to understand if a new session originated from a python handler has been actually answered by the remote user or by the operator voice mail with a message like "the customer you are trying to reach is not available at the moment" ? The python code looks something like the following: def handler(session, args): ## I do various things.... and than session_string = str("sofia/gateway/MY_GW/0039123456789?) new_session = freeswitch.Session(session_string) if new_session.ready(): # I get here also in case of automatic voice mail message else: # I supposed i?d get here in case of automatic ?unreachable? messages? Thanks in advance Lorenzo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/f52e7ce1/attachment.html From brian at freeswitch.org Fri Feb 26 02:35:01 2016 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Feb 2016 17:35:01 -0600 Subject: [Freeswitch-users] SRTP breaks my TLS session In-Reply-To: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> Message-ID: Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections. On Thu, Feb 25, 2016 at 5:13 PM, Emrah wrote: > Hello list, > I thought I had solved this issue by reducing my codec list to a minimum, > but it still persists, unfortunately. This was to reduce the TLS packet > size. > Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The > problem goes away when I disable SRTP. I only work over TLS. > Incoming calls work reliably with or without SRTP. > > How do you suggest debugging this? > I tried setting up a fresh instance of FS but the issue persists. > Now. It should be noted that calls fail sensibly more often when I have > more than one account registered on the same server with the same device. > > Any suggestion is welcome. Have you experienced this? > > I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is > a Yealink SIP-T46G running firmware 28.80.0.95. > > 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 > -- *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/20160225/5b0f09e2/attachment-0001.html From lmorley at neny.cslimits.net Fri Feb 26 03:09:26 2016 From: lmorley at neny.cslimits.net (Larry Morley) Date: Thu, 25 Feb 2016 19:09:26 -0500 Subject: [Freeswitch-users] Getting 503 when starting automatically In-Reply-To: References: Message-ID: David, What happens if you either don't start FS via systemd or stop it once the system's done booting, then start it like, say, su PATH=/usr/local/freeswitch/bin:$PATH freeswitch -nc Note that you'll likely want to keep a record of what the permissions on your FS files and directories are BEFORE doing this. Does it work then? Can you see from the console where the problem is? I'm nearly positive that I've chased this same error myself before. Unfortunately, I can't recall what the root cause or the fix was. Good luck! - Larry On Feb 25, 2016 16:30, wrote: > On Thu, 2016-02-25 at 21:26:09 +0300, Bote wrote: > > A 503 error, huh? Weird. Check for permissions problems. > > > My guess is that you are building as "root" so when you first ran it > > manually all the databases and log files were created with root as owner. > > But when it boots systemd is starting it as root and then dropping > > privileges to the freeswitch user as it should. After dropping > privileges it > > can't access its config files or open ports necessary to function. > > Thanks for the suggestion. I compiled as a normal user using Arch Linux's > makepkg tool and installed using its pacman tool, so I don't think that's > the problem. I'm also letting system start freeswitch.service with its > default user setting (as root). > > The weirdest thing is that it does start and connect fine using systemd > when make a request after boot to have systemd start it (systemctl start > freeswitch or systemctl restart freeswitch). In both cases, systemd is > controlling process identity, so I don't think it's a permissions issue. My > best guess is there's some kind of race condition at boot and if FreeSWITCH > starts too early, it gets a cached but invalid handle of some sort that > doesn't allow communicating in or out. This is just a guess though. > > Thanks, > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/3a18c08b/attachment.html From freeswitch at digitaldescent.net Fri Feb 26 04:18:15 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Thu, 25 Feb 2016 17:18:15 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: Moishe, Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls. It seems, I can only make outbound calls. Internal -> External is working. External -> Internal only works to the FS server, not to the extension. Internal -> Internal does not work. On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein wrote: > Does internal call also go direct to voicemail? > > Are you sure you don?t have a *99 before the extension number? > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Thursday, February 25, 2016 11:00 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Ok, in that case, what do I do next to figure out why inbound calls go > directly to voicemail instead of ringing the extension? > > On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: > > Ok, in that case, what do I look at next to see why it goes straight to > voicemail instead of ringing my extension? > > On Feb 25, 2016 5:52 AM, "Brian West" wrote: > > Its just the channel name so don't get tripped up by that, its just named > with some default data about the session. > > > > On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow < > freeswitch at digitaldescent.net> wrote: > > Debug console output: https://pastebin.freeswitch.org/24572 > > > > My fqdn only resolves inside the network, but the external-ip is > configured in FS to use stun, which reports my external ip. > > > > >Is your endpoint registered? How often is it set to reregister? > > Yes, and the zoiper default registration expiry is 3600 > > > > >Does your router have a sip alg? Are the ports opened in your firewall? > > > ish, no GUI for it, but I did make sure to disable it from the command > line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but > allow bi-directional with existing states. I do get 2 way audio when > calling out, and I also can hear the system forwarding my call in to > voicemail; the voicemail message and prompts. I know asterisk is different > from FS, but my asterisk experience has proven that at least with asterisk, > port forwarding isn't needed. Also, unfortunately, I cannot get a static > ip. > > > > Thanks! > > -Brian > > > > On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: > > It is probably sending call to your extension at domainname, if your > external ip is your domain name then you will see the external ip. > > Is your endpoint registered? How often is it set to reregister? > > Does your router have a sip alg? Are the ports opened in your firewall? > > Very hard to guess without a log of a call with debug enabled. > > sofia global sip trace on > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Wednesday, February 24, 2016 10:00 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Hello all! > > I'm new to freeswitch, so I'm sure this is just a newbie configuration > error. Sorry if it's been answered a million times, my google searches > always just bring up the standard NAT configuration pages. I've already > followed the confluence page on configuring NAT. > > > > My Setup: > > Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 > instance. > > SIP Provider: Flowroute > > > > NAT: extension and freeswitch are on the same network, both of which are > behind NAT and connecting to flowroute. I configured external sip and rtp > to use stun entries. External profile is using $${external_rtp/sip_ip} for > ext-rtp/sip respectively. > > > > I have one DID configured to go directly to my one extension. My > extension can register just fine. My extension can dial out. When I call > my cell, the call connects and I have 2 way audio. When I dial my DID from > my cell, I can see the call hitting the FS server, but instead of ringing > my extension, it goes straight to my extensions voicemail (which I can just > fine). > > > > When I look the at the console, it appears (sorry if this is wrong, I'm > only a day into free switch) that FS is attempting to route the call to my > extension...@ my external ip? > > > > I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] > switch_channel.c:1101 New Channel sofia/internal/@ > **:46072 > > > > > > Shouldn't that read sofia/internal/@ ? > > > > Sofia Status says: > > > > Name Type > Data State > > > ================================================================================================= > > external-ipv6 profile > sip:mod_sofia@[::1]:5080 RUNNING (0) > > external profile > sip:mod_sofia@:5080 RUNNING (0) > > external:: gateway sip:@ > sip.flowroute.com REGED > > internal-ipv6 profile > sip:mod_sofia@[::1]:5060 RUNNING (0) > > internal profile > sip:mod_sofia@:5060 RUNNING (0) > > > ================================================================================================= > > > > If I'm completely off base here, can anyone recommend where I can start > looking to change/troubleshoot the issue? I feel like it's just me missing > something, I just can't determine what that might be. > > > > Thanks, > > -Brian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/c205c604/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160225/c205c604/attachment-0001.jpg From max at nysolutions.com Fri Feb 26 04:50:28 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 26 Feb 2016 01:50:28 +0000 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> Message-ID: <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> Can you show a log of a internal call? Are the phones showing registered? Show registrations Did you by any chance add your ip range to the ACL. 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 Brian Chow Sent: Thursday, February 25, 2016 8:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Moishe, Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls. It seems, I can only make outbound calls. Internal -> External is working. External -> Internal only works to the FS server, not to the extension. Internal -> Internal does not work. On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein > wrote: Does internal call also go direct to voicemail? Are you sure you don?t have a *99 before the extension number? 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 Brian Chow Sent: Thursday, February 25, 2016 11:00 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension? On Feb 25, 2016 5:52 AM, "Brian West" > wrote: Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow > wrote: Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip. >Is your endpoint registered? How often is it set to reregister? Yes, and the zoiper default registration expiry is 3600 >Does your router have a sip alg? Are the ports opened in your firewall? > ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip. Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on 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 Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@:46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160226/c646cc1e/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/20160226/c646cc1e/attachment-0001.jpg From italo at freeswitch.org Fri Feb 26 05:00:49 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 25 Feb 2016 23:00:49 -0300 Subject: [Freeswitch-users] discriminate between session answered by users and voice mail automatic messages In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd On Thu, Feb 25, 2016 at 7:24 PM, sharigo roma wrote: > Dear freeswitchers, > > is there a way to understand if a new session originated from a python > handler has been actually answered by the remote user or by the operator > voice mail with a message like "the customer you are trying to reach is not > available at the moment" ? > > The python code looks something like the following: > > > def handler(session, args): > ## I do various things.... and than > > session_string = str("sofia/gateway/MY_GW/0039123456789?) > new_session = freeswitch.Session(session_string) > if new_session.ready(): > # I get here also in case of automatic voice mail message > else: > # I supposed i?d get here in case of automatic ?unreachable? messages? > > > Thanks in advance > Lorenzo > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/11446fed/attachment.html From italo at freeswitch.org Fri Feb 26 05:05:30 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Thu, 25 Feb 2016 23:05:30 -0300 Subject: [Freeswitch-users] mod_distributor / gwlist down In-Reply-To: References: Message-ID: you missed the profile keywork in your dialplan: sofia PROFILE internal internal_trk_1 gwlist down On Wed, Feb 24, 2016 at 1:38 PM, Dominique Jeannerod < dominique.jeannerod at interact-iv.com> wrote: > Hello, > > i'm currently working with mod_distributor, to load balance calls to > gateways. > The load balancing works well, and it's rather simple to configure, no > problem. > > I'm trying to automatically exclude down gateways from the load balancing, > which seems quite simple, too, when reading the mod_distributor doc... but > i can't make it work. > > I'm using freeswitch 1.6.6. > I configured a profile internal_trk_1, and 2 gateways (svc-isr-fr-301, and > non_existent_gw), with ping option. > One of the gateways, called (non_existent_gw) is down, and detected down > by sofia > > When I check the status of the gateways, the result is ok : > sofia profile internal_trk_1 gwlist down > non_existent_gw > > When I check with distributor command in the cli, it's also ok : > expand distributor I_MUT_SIPGTW ${sofia profile internal_trk_1 gwlist down} > svc-isr-fr-301 > > But I can't make it work inside the dialplan. I tested using the 2 > syntaxes listed in > https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor : > > > > > > The dead gateway still gets used by the load balancer. > > What I am doing wrong ? > > Best regards > > > Dominique Jeannerod > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160225/56522835/attachment.html From andrew.keil at visytel.com Fri Feb 26 05:33:08 2016 From: andrew.keil at visytel.com (Andrew Keil) Date: Fri, 26 Feb 2016 02:33:08 +0000 Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function Message-ID: To FreeSWITCH Users, See below for a sample template for a Lua Service Script running inside FreeSWITCH. The issue I have is fairly straightforward. I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script. This function is CleanUp(). Then I would like the service to end. The problem I am having is if the caller hangs up during the playback of "intro.wav" (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()). What actually happens is it returns to MainService() and continues to try and play "info.wav", unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call. My aim is to reduce extra code and to make the Lua script simpler and easier to read. Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc.... So is there a way to stop a Lua script running inside FreeSWITCH cleanly? I have tried the os.exit() this is barred from use by FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution. FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use "if (session:ready()) then ...." or "if (not session:ready()) then goto HANGUPEXIT end" makes the code ugly. Thanks in advance, Andrew Keil Visytel Pty Ltd ------------------------------------------------------------------------------ Sample Lua Service ----------------------------------------------------------------------- -- Lua template for FreeSWITCH service -- By: Andrew Keil (Visytel Pty Ltd) -- Email: support at visytel.com -- Setup script wide variables here function PreAnswer() freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); -- Add your pre answer code from here -- End of your pre answer code freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n"); end function AnswerCaller() session:answer() session:sleep(1000) end function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n"); if (session:ready()) then -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end -- Add your main service code from here (caller would have been answered) session:streamFile("intro.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("info.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("outro.wav") if (not session:ready()) then goto HANGUPEXIT end -- End of your main service code end ::ENDSERVICE:: if (session:ready()) then -- End of service so hangup session:hangup() -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage end goto END ::HANGUPEXIT:: freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n"); ::END:: freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n"); end function CleanUp() freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); -- Add your cleanup code from here (caller would have been disconnected) -- End of your cleanup code freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n"); end function myHangupHook(s, status, arg) session:hangup() CleanUp() -- Run CleanUp function now since the caller has disconnected end -- Setup Hangup event handler here v_hangup = "HANGUP" session:setHangupHook("myHangupHook", "v_hangup") -- Call service functions in order PreAnswer() AnswerCaller() MainService() -- End of Lua service -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/e47a8034/attachment-0001.html From mario_fs at mgtech.com Fri Feb 26 08:37:41 2016 From: mario_fs at mgtech.com (Mario G) Date: Thu, 25 Feb 2016 21:37:41 -0800 Subject: [Freeswitch-users] SRTP breaks my TLS session In-Reply-To: References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> Message-ID: <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com> You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export. > On Feb 25, 2016, at 3:35 PM, Brian West wrote: > > Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections. > > On Thu, Feb 25, 2016 at 5:13 PM, Emrah > wrote: > Hello list, > I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size. > Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS. > Incoming calls work reliably with or without SRTP. > > How do you suggest debugging this? > I tried setting up a fresh instance of FS but the issue persists. > Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device. > > Any suggestion is welcome. Have you experienced this? > > I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95. > > 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 > > > > -- > 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/20160225/1c4ba132/attachment.html From lists at kavun.ch Fri Feb 26 12:36:49 2016 From: lists at kavun.ch (Emrah) Date: Fri, 26 Feb 2016 10:36:49 +0100 Subject: [Freeswitch-users] SRTP breaks my TLS session In-Reply-To: <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com> References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com> Message-ID: <93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch> Thanks for this. This isn?t just a yealink thing. I?ve encountered sporadic issues with soft phones and other desk phones as well. I didn?t use the PCap capture feature because I had assumed it would give me a bunch of TLS packets. I?ll test that and revert back. How can we explain that I have more calls failing if I register multiple accounts? Emrah > On Feb 26, 2016, at 6:37 AM, Mario G wrote: > > You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export. > >> On Feb 25, 2016, at 3:35 PM, Brian West > wrote: >> >> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections. >> >> On Thu, Feb 25, 2016 at 5:13 PM, Emrah > wrote: >> Hello list, >> I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size. >> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS. >> Incoming calls work reliably with or without SRTP. >> >> How do you suggest debugging this? >> I tried setting up a fresh instance of FS but the issue persists. >> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device. >> >> Any suggestion is welcome. Have you experienced this? >> >> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95. >> >> 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 >> >> >> >> -- >> 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/20160226/77488862/attachment-0001.html From tarantul at gmail.com Fri Feb 26 13:04:34 2016 From: tarantul at gmail.com (Nick 'tarantul' Novikov) Date: Fri, 26 Feb 2016 13:04:34 +0300 Subject: [Freeswitch-users] no audio after hold In-Reply-To: References: Message-ID: Do you have any news about this bug? Can I help? On Wed, Feb 17, 2016 at 11:32 PM, Nick 'tarantul' Novikov < tarantul at gmail.com> wrote: > Hello! > > Done. > https://freeswitch.org/jira/browse/FS-8840 > > On Wed, Feb 17, 2016 at 5:31 PM, Brian West wrote: > >> bug reports go on JIRA please. >> >> On Wed, Feb 17, 2016 at 4:03 AM, Nick 'tarantul' Novikov < >> tarantul at gmail.com> wrote: >> >>> Hello! >>> >>> Log file is large, I can't paste it in FS pastebin. >>> You may download gzip log file here >>> https://yadi.sk/d/pLbEHi_LoyWsS >>> >>> On Tue, Feb 16, 2016 at 7:11 PM, Brian West >>> wrote: >>> >>>> Your logs are incomplete, need full debug logs. Please don't filter >>>> them excessively and try on master please. >>>> >>>> On Tue, Feb 16, 2016 at 10:06 AM, Giovanni Maruzzelli < >>>> gmaruzz at gmail.com> wrote: >>>> >>>>> Please pastebin a complete sip trace (from fs-cli: sofia global >>>>> siptrace on) and then put here the link to pastebin. >>>>> Il 16/Feb/2016 16:29, "Nick 'tarantul' Novikov" >>>>> ha scritto: >>>>> >>>>>> Hello >>>>>> >>>>>> I have some problem with sound after hold usage. Sometime the sound >>>>>> disappear after callcenter agent shift unhold.For callcenter agents we use >>>>>> sip.js (version 0.7.2). >>>>>> In freeswitch logs I see strange lines ( >>>>>> https://pastebin.freeswitch.org/24557). >>>>>> Can anyone explain, what happened with audio and how to fix it? >>>>>> >>>>>> >>>>>> -- >>>>>> tarantul >>>>>> Dios es Amor >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 >>>> >>> >>> >>> >>> -- >>> tarantul >>> Dios es Amor >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> > > > > -- > tarantul > Dios es Amor > -- tarantul Dios es Amor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/b030e7f3/attachment.html From lists at kavun.ch Fri Feb 26 14:24:25 2016 From: lists at kavun.ch (Emrah) Date: Fri, 26 Feb 2016 12:24:25 +0100 Subject: [Freeswitch-users] Joining several FS servers in different regions In-Reply-To: <20160224152350.GA30297@blomma.liberationtech.net> References: <20160224152350.GA30297@blomma.liberationtech.net> Message-ID: <50E27BAC-AA7E-4E20-88CC-648CD0DEC6D0@kavun.ch> Hi there, I was recently confronted to a similar challenge. In my case, there was an additional use case where Bob could have multiple phones registered simultaneously both on the North American server and the EU server. I strive to involve as little components as possible to avoid unnecessary breaking points. So what I came up with is a very basic but working solution. I modified my dial string so that user/1000 at domain would also ring the alternative server in bypass media mode. I also share the same directory files and dial plan files that I host on a centralized storage space. My gateways are set up strategically to route the EU server to the EU POPs and the U.S. server to the NA POPs. That is for my providers who have multiple POPs distributed geographically. With the set up as described here, both servers are active and passive at the same time in the cluster. And it can easily be expanded. To my knowledge, it?s as optimized as it can get. Best, Emrah > On Feb 24, 2016, at 4:23 PM, Oivvio Polite wrote: > > > I'm working on a Saas service that will connect WebRTC (with SIP or Verto for > signaling) clients to one another and to PSTN. I'm aware of other such > services, but this is tailored to a specific niche. I also want WebRTC > media to flow through FS, rather than P2P, so that I can record it. > > The primary markets are Europe and North America. To keep latency low I need > to have servers on both continents. Later on I might want to add more > regions. I'm not too worried about high availability or high loads. > > The clients can keep track of which FS is closest to them and register > with that. But this still leaves me with a couple of questions. > > > > 1. Maintaining a single user database > ===================================== > > What's the simplest way of maintaining one joint user database? (User > directory in FS parlance). > > The first thing that comes to mind is to generate the xml dynamically > from a central database, and force as `reloadxml` on all servers every > time there's an update. The communication between central database and > FS servers could be via RabbitMQ. mod_xml_curl could also be a part of > this I guess. > > How does that sound? Is there an other simpler way of achiving my goal > that I'm not seeing? > > > > 2. Keeping track of where a user i currently registered > ======================================================= > > Let's say I have a client Alice who's registered with a FS server in > North America and a client Bob who has registered wit a FS server in Europe. > > Now Alice tries to call Bob. How does the North America server know that > Bob is currently registered with the Europe server and that the call > should be routed through there? > > For this second problem I don't even have a tentative solution so I'm > curious to hear any ideas. > > 3. SIP Trunks and geography > =========================== > > If Alice is in North America and wants to call a PSTN endpoint in North > America I should obviously be using a SIP trunk in North America, since > going over the Atlantic twice will add a lot of latency. > > But what about when Alice want to call a PSTN endpoint in Europe? Is it > better to use a SIP endpoint in Europe or in North America? > > Thanks in advance Oivvio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Alexey.Gorbachev at leomax.ru Fri Feb 26 12:47:47 2016 From: Alexey.Gorbachev at leomax.ru (Gorbachev Alexey) Date: Fri, 26 Feb 2016 09:47:47 +0000 Subject: [Freeswitch-users] FreeSwitch second recovery Message-ID: I installed FreeSwitch cluster according to official manual -https://wiki.freeswitch.org/wiki/Freeswitch_HA And it works, when I power off first node current calls successfully move to second node and voice disappears only for 3 second. The problem is when I power on first node, server starts FreeSwitch, and FreeSwitch during start up clears calls in database and of cause I can't move current calls back again to first node. Can I move current calls between servers without interruption again? Thank you. Best regards, Gorbachev Alexey, System administrator Linux LEOMAX Group Tel: +7(495)967-7797, ext. 4119, Cell: +7(926)982-9520 Mail to: alexey.gorbachev at leomax.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/7c30b749/attachment-0001.html From Adam.Seeliger at qsc.de Fri Feb 26 15:37:07 2016 From: Adam.Seeliger at qsc.de (Seeliger, Adam) Date: Fri, 26 Feb 2016 12:37:07 +0000 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> Hi and thanks for the feedback, sry that I did not respond for a long time. I already use: I also tested all mentioned params below, nothing works. When I register a User via TLS FreeSWITCH does not even try to ping the user. I turned sofia global siptrace on and watched the flow: User Server 13:09:33.311446: REGISTER [TLS] -> 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] 13:09:33.331948: REGISTER (AUTH) [TLS] -> 13:09:33.336959: <- 200 OK [TLS] Nothing happens 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered. Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration? Thanks in advance, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah Gesendet: Freitag, 29. Januar 2016 09:25 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Hi! This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS. I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set. I found out that in my case, what works best even with TLS, is to use: This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports. Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead. You could also look into: --> --> I?ll leave it up to you to investigate those options more in details on the FS documentation. Please keep us posted! E On Jan 28, 2016, at 11:48 AM, Seeliger, Adam > wrote: Hi all, I have a problem, when I enable TLS and register a phone using TLS on Port 5061. FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone. Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?. I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS. Is there any way to ping TLS registered using TLS? Thanks in advance - Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160226/31ad65c0/attachment-0001.html From s.safarov at gmail.com Fri Feb 26 15:58:41 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 26 Feb 2016 15:58:41 +0300 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: If your phone has enabled SIPS uri please disable and use sip+tls. On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam wrote: > Hi and thanks for the feedback, > > > > sry that I did not respond for a long time. > > > > I already use: > > value="NDLB-connectile-dysfunction-2.0"/> > > > > I also tested all mentioned params below, nothing works. > > > > When I register a User via TLS FreeSWITCH does not even try to ping the > user. > > I turned sofia global siptrace on and watched the flow: > > > > User Server > > 13:09:33.311446: REGISTER [TLS] -> > > 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] > > 13:09:33.331948: REGISTER (AUTH) [TLS] -> > > 13:09:33.336959: <- 200 OK [TLS] > > Nothing happens > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is > now Unreachable > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user > 'user at host' due to options failure > > > > When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the > user as soon as he is registered. > > > > Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, > depending on the registration? > > > > Thanks in advance, > > Adam > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah > *Gesendet:* Freitag, 29. Januar 2016 09:25 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > Hi! > > This is interesting. I experienced something rather similar where calls > would drop because FS would timeout on certain packets sent over UDP > instead of TLS. > > I assume you mean FS exits with port 5060 instead of port 5061? Because > the port on the remote end should be dynamically set. > > I found out that in my case, what works best even with TLS, is to use: > > name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/> > > This goes as far as it can to lay out the path to contacting the client > with all consideration in regards to NAT and dynamic ports. > > Not sure if it will help you. I?ve personally disabled options-ping an let > my clients deal with keep-alive instead. > > > > You could also look into: > > > > --> > > --> > > > > > > > > I?ll leave it up to you to investigate those options more in details on > the FS documentation. > > > > Please keep us posted! > > > > E > > On Jan 28, 2016, at 11:48 AM, Seeliger, Adam wrote: > > > > Hi all, > > > > I have a problem, when I enable TLS and register a phone using TLS on Port > 5061. > > FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, > which is ignored by the phone. > > Moments later FreeSWITCH deletes the registration, because > ?unregister-on-options-fail? is set to ?true?. > > > > I already figured out, that you can set ?all-reg-options-ping? to > ?udp-only?, but this would completely disable this feature for TLS. > > Is there any way to ping TLS registered using TLS? > > > > Thanks in advance > > > > - Adam > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/c35e68d5/attachment.html From Adam.Seeliger at qsc.de Fri Feb 26 16:21:36 2016 From: Adam.Seeliger at qsc.de (Seeliger, Adam) Date: Fri, 26 Feb 2016 13:21:36 +0000 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> Hi, the phone uses sip+tls. I test using a snom715, got plenty other phones here, but I guess they will behave the same way. It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls) Regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 13:59 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue If your phone has enabled SIPS uri please disable and use sip+tls. On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: Hi and thanks for the feedback, sry that I did not respond for a long time. I already use: I also tested all mentioned params below, nothing works. When I register a User via TLS FreeSWITCH does not even try to ping the user. I turned sofia global siptrace on and watched the flow: User Server 13:09:33.311446: REGISTER [TLS] -> 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] 13:09:33.331948: REGISTER (AUTH) [TLS] -> 13:09:33.336959: <- 200 OK [TLS] Nothing happens 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered. Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration? Thanks in advance, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah Gesendet: Freitag, 29. Januar 2016 09:25 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Hi! This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS. I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set. I found out that in my case, what works best even with TLS, is to use: This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports. Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead. You could also look into: --> --> I?ll leave it up to you to investigate those options more in details on the FS documentation. Please keep us posted! E On Jan 28, 2016, at 11:48 AM, Seeliger, Adam > wrote: Hi all, I have a problem, when I enable TLS and register a phone using TLS on Port 5061. FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone. Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?. I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS. Is there any way to ping TLS registered using TLS? Thanks in advance - Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160226/d2df6840/attachment-0001.html From telishisheer at gmail.com Fri Feb 26 16:39:03 2016 From: telishisheer at gmail.com (Shisheer Teli) Date: Fri, 26 Feb 2016 19:09:03 +0530 Subject: [Freeswitch-users] Multiple device registration on single extension Message-ID: Dear Team, Multiple device registrations on a single extension are not working. I made changes in internal profile as "" . but still not working. I am using FreeSWITCH (Version 1.6.5 git 64bit) on Linux -- Regards, Shisheer T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/149d629e/attachment.html From mike at jerris.com Fri Feb 26 16:49:40 2016 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Feb 2016 08:49:40 -0500 Subject: [Freeswitch-users] Multiple device registration on single extension In-Reply-To: References: Message-ID: Can you give us any more information? On Friday, February 26, 2016, Shisheer Teli wrote: > Dear Team, > > Multiple device registrations on a single extension are not working. > > I made changes in internal profile as " name="multiple-registrations" value="true"/>" . > > but still not working. > > I am using FreeSWITCH (Version 1.6.5 git 64bit) on Linux > > > -- > Regards, > Shisheer T > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/2b3a861b/attachment.html From Chris.Young at enghouse.com Fri Feb 26 17:20:00 2016 From: Chris.Young at enghouse.com (Chris Young) Date: Fri, 26 Feb 2016 14:20:00 +0000 Subject: [Freeswitch-users] Call gets disconnected on write frame error Message-ID: Hi all, Recently, we've been experiencing a problem whereby FreeSWITCH disconnects a call after receiving a frame of audio data with a data length of 0. When this happens we see the following messages in the log: [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 bytes to accommodate 0->320 [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed! and the call gets disconnected. This normally happens when trying to join a conference but we have also seen it when bridging two calls together. Is the call disconnection by-design or just a side-effect of having received the bad frame of data? Thanks, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/1ba15d77/attachment.html From abaci64 at gmail.com Fri Feb 26 17:29:17 2016 From: abaci64 at gmail.com (Abaci B) Date: Fri, 26 Feb 2016 09:29:17 -0500 Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function In-Reply-To: References: Message-ID: See https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook for a few ways to exit the lua script (error(), return "exit", return "die", s:destroy("error message")). I personally tried return "exit" but it seems to me that it only exits the calling function, haven't had a chance to look further, it's possible that the calling it from the within a function is different. if you play around and figure out please report back. On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil wrote: > To FreeSWITCH Users, > > > > See below for a sample template for a Lua Service Script running inside > FreeSWITCH. > > > > The issue I have is fairly straightforward. > > > > I need a function to run when hangup is detected (ie. at the end of the > call) however I understand this must not delay ending the script. This > function is CleanUp(). Then I would like the service to end. > > > > The problem I am having is if the caller hangs up during the playback of > ?intro.wav? (as shown inside the MainService() function below), then the > code jumps to the myHangupHook which calls CleanUp() perfectly, the issue > is once CleanUp() is complete I would like the Lua script to end there and > then (ie. at the bottom of CleanUp()). What actually happens is it returns > to MainService() and continues to try and play ?info.wav?, unless I either > check for session:ready() everywhere or add a goto as shown below under > each streamFile() function call. > > > > My aim is to reduce extra code and to make the Lua script simpler and > easier to read. Also I would like to try and avoid goto statements, which > I know can be done with if (session:ready()) etc?. > > > > So is there a way to stop a Lua script running inside FreeSWITCH cleanly? > I have tried the os.exit() this is barred from use by FreeSWITCH. I have > also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on > CentOS 6.7, CentOS 7 and windows) 100% of the time! > > > > I could look further into the Lua additions done by the FreeSWITCH team in > the source code, however if someone has already solved this then that would > be the best solution. > > > > FYI: Obviously the script below is simple, however I am sure that you > understand if the script was complicated having to use ?*if > (session:ready()) then ?.?* or ?*if (not session:ready()) then goto > HANGUPEXIT end?* makes the code ugly. > > > > Thanks in advance, > > > > Andrew Keil > > *Visytel Pty Ltd* > > > > > > > > ------------------------------------------------------------------------------ > Sample Lua Service > ----------------------------------------------------------------------- > > > > -- Lua template for FreeSWITCH service > > -- By: Andrew Keil (Visytel Pty Ltd) > > -- Email: support at visytel.com > > > > -- Setup script wide variables here > > > > function PreAnswer() > > freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); > > -- Add your pre answer code from here > > > > -- End of your pre answer code > > freeswitch.consoleLog("INFO", "PRE ANSWER SECTION > COMPLETE\n"); > > end > > > > function AnswerCaller() > > session:answer() > > session:sleep(1000) > > end > > > > function MainService() > > freeswitch.consoleLog("INFO", "MAIN SERVICE > SECTION\n"); > > if (session:ready()) then > > -- Note (1): If you wish to end the call > then simply use: goto ENDSERVICE > > -- Note (2): To terminate the service > sooner when HANGUP is detected use: if (not session:ready()) then goto > HANGUPEXIT end > > -- Add your main service code from > here (caller would have been answered) > > > > session:streamFile("intro.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > session:streamFile("info.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > > session:streamFile("outro.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > > > -- End of your main service code > > end > > ::ENDSERVICE:: > > if (session:ready()) then > > -- End of service so hangup > > session:hangup() -- Should automatically > jump to CleanUp() via hangup handler if caller still online at this stage > > end > > goto END > > ::HANGUPEXIT:: > > freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP > DETECTED)\n"); > > ::END:: > > freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION > COMPLETE\n"); > > end > > > > function CleanUp() > > freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); > > -- Add your cleanup code from here (caller would have been > disconnected) > > > > -- End of your cleanup code > > freeswitch.consoleLog("INFO", "CLEANUP SECTION > COMPLETE\n"); > > end > > > > function myHangupHook(s, status, arg) > > session:hangup() > > CleanUp() -- Run CleanUp function now since the caller has > disconnected > > end > > > > -- Setup Hangup event handler here > > v_hangup = "HANGUP" > > session:setHangupHook("myHangupHook", "v_hangup") > > > > -- Call service functions in order > > PreAnswer() > > AnswerCaller() > > MainService() > > -- End of Lua service > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/c27bd9fb/attachment-0001.html From s.safarov at gmail.com Fri Feb 26 17:44:53 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 26 Feb 2016 17:44:53 +0300 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: Please send output of command "sofia status profile internal reg " On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam wrote: > Hi, > > > > the phone uses sip+tls. > > I test using a snom715, got plenty other phones here, but I guess they > will behave the same way. > > It really looks like FreeSWITCH is doing something wrong (or is wrongly > configured ? if there are any parameters for options ping supporting both, > udp and tls) > > > > Regards > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey > Safarov > *Gesendet:* Freitag, 26. Februar 2016 13:59 > > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > If your phone has enabled SIPS uri please disable and use sip+tls. > > > > > > On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: > > Hi and thanks for the feedback, > > > > sry that I did not respond for a long time. > > > > I already use: > > value="NDLB-connectile-dysfunction-2.0"/> > > > > I also tested all mentioned params below, nothing works. > > > > When I register a User via TLS FreeSWITCH does not even try to ping the > user. > > I turned sofia global siptrace on and watched the flow: > > > > User Server > > 13:09:33.311446: REGISTER [TLS] -> > > 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] > > 13:09:33.331948: REGISTER (AUTH) [TLS] -> > > 13:09:33.336959: <- 200 OK [TLS] > > Nothing happens > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is > now Unreachable > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user > 'user at host' due to options failure > > > > When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the > user as soon as he is registered. > > > > Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, > depending on the registration? > > > > Thanks in advance, > > Adam > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah > *Gesendet:* Freitag, 29. Januar 2016 09:25 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > Hi! > > This is interesting. I experienced something rather similar where calls > would drop because FS would timeout on certain packets sent over UDP > instead of TLS. > > I assume you mean FS exits with port 5060 instead of port 5061? Because > the port on the remote end should be dynamically set. > > I found out that in my case, what works best even with TLS, is to use: > > name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/> > > This goes as far as it can to lay out the path to contacting the client > with all consideration in regards to NAT and dynamic ports. > > Not sure if it will help you. I?ve personally disabled options-ping an let > my clients deal with keep-alive instead. > > > > You could also look into: > > > > --> > > --> > > > > > > > > I?ll leave it up to you to investigate those options more in details on > the FS documentation. > > > > Please keep us posted! > > > > E > > On Jan 28, 2016, at 11:48 AM, Seeliger, Adam wrote: > > > > Hi all, > > > > I have a problem, when I enable TLS and register a phone using TLS on Port > 5061. > > FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, > which is ignored by the phone. > > Moments later FreeSWITCH deletes the registration, because > ?unregister-on-options-fail? is set to ?true?. > > > > I already figured out, that you can set ?all-reg-options-ping? to > ?udp-only?, but this would completely disable this feature for TLS. > > Is there any way to ping TLS registered using TLS? > > > > Thanks in advance > > > > - Adam > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/0d9a7080/attachment-0001.html From Adam.Seeliger at qsc.de Fri Feb 26 17:59:14 2016 From: Adam.Seeliger at qsc.de (Seeliger, Adam) Date: Fri, 26 Feb 2016 14:59:14 +0000 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> Hi, here is the requested output. I changed the real user, domain and ip address values into descriptions. I guess the values are not necessary? Both, FreeSWITCH and the phone are in the same network (no NAT involved here) Registrations: ================================================================================================= Call-ID: 3134353634383833373032323631-ncwgvit2obfp User: user at domain Contact: "User Name" Agent: snom715/8.7.5.35 Status: Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912) Ping-Status: Reachable Host: hostname IP: ip Port: 60206 Auth-User: user Auth-Realm: domain MWI-Account: user at domain Total items returned: 1 ================================================================================================= 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure As you can see, the user immediately got unregistered again ? Best regards, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 15:45 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Please send output of command "sofia status profile internal reg " On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam > wrote: Hi, the phone uses sip+tls. I test using a snom715, got plenty other phones here, but I guess they will behave the same way. It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls) Regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 13:59 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue If your phone has enabled SIPS uri please disable and use sip+tls. On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: Hi and thanks for the feedback, sry that I did not respond for a long time. I already use: I also tested all mentioned params below, nothing works. When I register a User via TLS FreeSWITCH does not even try to ping the user. I turned sofia global siptrace on and watched the flow: User Server 13:09:33.311446: REGISTER [TLS] -> 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] 13:09:33.331948: REGISTER (AUTH) [TLS] -> 13:09:33.336959: <- 200 OK [TLS] Nothing happens 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered. Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration? Thanks in advance, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah Gesendet: Freitag, 29. Januar 2016 09:25 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Hi! This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS. I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set. I found out that in my case, what works best even with TLS, is to use: This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports. Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead. You could also look into: --> --> I?ll leave it up to you to investigate those options more in details on the FS documentation. Please keep us posted! E On Jan 28, 2016, at 11:48 AM, Seeliger, Adam > wrote: Hi all, I have a problem, when I enable TLS and register a phone using TLS on Port 5061. FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone. Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?. I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS. Is there any way to ping TLS registered using TLS? Thanks in advance - Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160226/4adb3c20/attachment-0001.html From brian at freeswitch.org Fri Feb 26 18:15:04 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Feb 2016 09:15:04 -0600 Subject: [Freeswitch-users] Call gets disconnected on write frame error In-Reply-To: References: Message-ID: What revision of FreeSWITCH are you running? On Fri, Feb 26, 2016 at 8:20 AM, Chris Young wrote: > Hi all, > > > > Recently, we've been experiencing a problem whereby FreeSWITCH disconnects > a call after receiving a frame of audio data with a data length of 0. When > this happens we see the following messages in the log: > > > > [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 > bytes to accommodate 0->320 > > [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed! > > > > and the call gets disconnected. This normally happens when trying to join > a conference but we have also seen it when bridging two calls together. > > > > Is the call disconnection by-design or just a side-effect of having > received the bad frame of data? > > > > Thanks, > > Chris > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/625cfb44/attachment.html From Chris.Young at enghouse.com Fri Feb 26 18:25:25 2016 From: Chris.Young at enghouse.com (Chris Young) Date: Fri, 26 Feb 2016 15:25:25 +0000 Subject: [Freeswitch-users] Call gets disconnected on write frame error In-Reply-To: References: Message-ID: <4a445ea4ec26444a986cf29ba2bc20c4@UK-MAIL-001.edge.local> 1.4.18. I realise that this is not the most up-to-date version but I'm not in a position to re-test with a newer release at the moment. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 26 February 2016 15:15 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call gets disconnected on write frame error What revision of FreeSWITCH are you running? On Fri, Feb 26, 2016 at 8:20 AM, Chris Young > wrote: Hi all, Recently, we've been experiencing a problem whereby FreeSWITCH disconnects a call after receiving a frame of audio data with a data length of 0. When this happens we see the following messages in the log: [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 bytes to accommodate 0->320 [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed! and the call gets disconnected. This normally happens when trying to join a conference but we have also seen it when bridging two calls together. Is the call disconnection by-design or just a side-effect of having received the bad frame of data? Thanks, Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160226/5477d7e4/attachment.html From s.safarov at gmail.com Fri Feb 26 18:52:30 2016 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 26 Feb 2016 15:52:30 +0000 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: Registration is correct. Think it bug and requred to fill a jira tiket. I use master 4 mouth old and it work correctly. On Fri, Feb 26, 2016, 18:00 Seeliger, Adam wrote: > Hi, here is the requested output. > > > > I changed the real user, domain and ip address values into descriptions. > > I guess the values are not necessary? > > > > Both, FreeSWITCH and the phone are in the same network (no NAT involved > here) > > > > Registrations: > > > ================================================================================================= > > Call-ID: 3134353634383833373032323631-ncwgvit2obfp > > User: user at domain > > Contact: "User Name" :60206;transport=tls;line=gvg9q8jh;fs_nat=yes;fs_path=sip%3Auser%40ip%3A60206> > > Agent: snom715/8.7.5.35 > > Status: Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) > EXPSECS(1912) > > Ping-Status: Reachable > > Host: hostname > > IP: ip > > Port: 60206 > > Auth-User: user > > Auth-Realm: domain > > MWI-Account: user at domain > > > > Total items returned: 1 > > > ================================================================================================= > > 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' > is now Unreachable > > 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user > 'user at domain' due to options failure > > > > As you can see, the user immediately got unregistered again L > > > > Best regards, > > Adam > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey > Safarov > *Gesendet:* Freitag, 26. Februar 2016 15:45 > > > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > Please send output of command "sofia status profile internal reg > " > > > > On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam > wrote: > > Hi, > > > > the phone uses sip+tls. > > I test using a snom715, got plenty other phones here, but I guess they > will behave the same way. > > It really looks like FreeSWITCH is doing something wrong (or is wrongly > configured ? if there are any parameters for options ping supporting both, > udp and tls) > > > > Regards > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Sergey > Safarov > *Gesendet:* Freitag, 26. Februar 2016 13:59 > > > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > If your phone has enabled SIPS uri please disable and use sip+tls. > > > > > > On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: > > Hi and thanks for the feedback, > > > > sry that I did not respond for a long time. > > > > I already use: > > value="NDLB-connectile-dysfunction-2.0"/> > > > > I also tested all mentioned params below, nothing works. > > > > When I register a User via TLS FreeSWITCH does not even try to ping the > user. > > I turned sofia global siptrace on and watched the flow: > > > > User Server > > 13:09:33.311446: REGISTER [TLS] -> > > 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] > > 13:09:33.331948: REGISTER (AUTH) [TLS] -> > > 13:09:33.336959: <- 200 OK [TLS] > > Nothing happens > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is > now Unreachable > > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user > 'user at host' due to options failure > > > > When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the > user as soon as he is registered. > > > > Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, > depending on the registration? > > > > Thanks in advance, > > Adam > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Emrah > *Gesendet:* Freitag, 29. Januar 2016 09:25 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] all-reg-options-ping and tls issue > > > > Hi! > > This is interesting. I experienced something rather similar where calls > would drop because FS would timeout on certain packets sent over UDP > instead of TLS. > > I assume you mean FS exits with port 5060 instead of port 5061? Because > the port on the remote end should be dynamically set. > > I found out that in my case, what works best even with TLS, is to use: > > name=?sip-force-contact? value="NDLB-connectile-dysfunction-2.0"/> > > This goes as far as it can to lay out the path to contacting the client > with all consideration in regards to NAT and dynamic ports. > > Not sure if it will help you. I?ve personally disabled options-ping an let > my clients deal with keep-alive instead. > > > > You could also look into: > > > > --> > > --> > > > > > > > > I?ll leave it up to you to investigate those options more in details on > the FS documentation. > > > > Please keep us posted! > > > > E > > On Jan 28, 2016, at 11:48 AM, Seeliger, Adam wrote: > > > > Hi all, > > > > I have a problem, when I enable TLS and register a phone using TLS on Port > 5061. > > FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, > which is ignored by the phone. > > Moments later FreeSWITCH deletes the registration, because > ?unregister-on-options-fail? is set to ?true?. > > > > I already figured out, that you can set ?all-reg-options-ping? to > ?udp-only?, but this would completely disable this feature for TLS. > > Is there any way to ping TLS registered using TLS? > > > > Thanks in advance > > > > - Adam > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/b19dcd24/attachment-0001.html From olegstolyar at gmail.com Fri Feb 26 20:38:34 2016 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 26 Feb 2016 09:38:34 -0800 Subject: [Freeswitch-users] Transcoding and recording Message-ID: Hi guys, if I record a session where both A and B use the same codec, will transcoding occur? Basically, in this case will the same codecs gain any CPU advantage? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/f49b38fb/attachment.html From brian at freeswitch.org Fri Feb 26 21:02:17 2016 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Feb 2016 12:02:17 -0600 Subject: [Freeswitch-users] Transcoding and recording In-Reply-To: References: Message-ID: Yes, transcoding is required to record a mux'ed stream or a stereo stream into a wav file. On Fri, Feb 26, 2016 at 11:38 AM, Oleg Stolyar wrote: > Hi guys, > > if I record a session where both A and B use the same codec, will > transcoding occur? > > Basically, in this case will the same codecs gain any CPU advantage? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/2f91650b/attachment.html From olegstolyar at gmail.com Fri Feb 26 21:21:21 2016 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 26 Feb 2016 10:21:21 -0800 Subject: [Freeswitch-users] Transcoding and recording In-Reply-To: References: Message-ID: Thanks Brian! On Fri, Feb 26, 2016 at 10:02 AM, Brian West wrote: > Yes, transcoding is required to record a mux'ed stream or a stereo stream > into a wav file. > > On Fri, Feb 26, 2016 at 11:38 AM, Oleg Stolyar > wrote: > >> Hi guys, >> >> if I record a session where both A and B use the same codec, will >> transcoding occur? >> >> Basically, in this case will the same codecs gain any CPU advantage? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160226/bda01ef0/attachment.html From blasterjr at gmail.com Fri Feb 26 21:28:47 2016 From: blasterjr at gmail.com (Chris Tunbridge) Date: Fri, 26 Feb 2016 11:28:47 -0700 Subject: [Freeswitch-users] How to kill call/channel on the reception of CUSTOM avmd::beep In-Reply-To: References: Message-ID: Glad to hear and i'm glad i could help. On Thu, Feb 25, 2016 at 2:15 PM, Aqs Younas wrote: > My fault. Got it working with uuid_kill Unique-ID > > Thanks. > > On 26 February 2016 at 01:01, Aqs Younas wrote: > >> Thanks for your Reply. >> >> freeswitch at debian> uuid_hangup Unique-ID >> Unknown Command: uuid_hangup 11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc >> >> Tried to kill. >> >> freeswitch at debian> uuid_kill Unique-ID >> >> -ERR No such channel! >> >> Any suggestion? >> >> On 25 February 2016 at 23:43, Chris Tunbridge >> wrote: >> >>> uuid_hangup the value of the Unique-ID field possibly? >>> >>> On Thu, Feb 25, 2016 at 10:13 AM, Aqs Younas >>> wrote: >>> >>>> Hi, >>>> >>>> I have subscribed to CUSTOM avmd::beep and this is what I get. >>>> >>>> "Event-Subclass": "avmd::beep", >>>> "Event-Name": "CUSTOM", >>>> "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", >>>> "FreeSWITCH-Hostname": "debian", >>>> "FreeSWITCH-Switchname": "debian", >>>> "FreeSWITCH-IPv4": "192.168.10.59", >>>> "FreeSWITCH-IPv6": "::1", >>>> "Event-Date-Local": "2016-02-25 10:46:17", >>>> "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", >>>> "Event-Date-Timestamp": "1456415177487145", >>>> "Event-Calling-File": "mod_avmd.c", >>>> "Event-Calling-Function": "avmd_process", >>>> "Event-Calling-Line-Number": "550", >>>> "Event-Sequence": "3174", >>>> "Beep-Status": "stop", >>>> "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", >>>> "call-command": "avmd" >>>> >>>> Now I need to hangup that call(beep detected), but could not find any >>>> event variable connected to the call. >>>> >>>> How can i kill the call. >>>> >>>> Any pointer is much appreciated. >>>> >>>> This email has been sent from a virus-free computer protected by Avast. >>>> www.avast.com >>>> >>>> <#-2020178404_996209302_-2019354057_-1458571440_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 >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/45175636/attachment-0001.html From mario_fs at mgtech.com Fri Feb 26 22:21:57 2016 From: mario_fs at mgtech.com (Mario G) Date: Fri, 26 Feb 2016 11:21:57 -0800 Subject: [Freeswitch-users] major build change in master. In-Reply-To: <8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com> References: <8C3827C6-22F2-45BB-BFAA-6AB64BD67E12@mgtech.com> <01e201d16ffb$db648f10$922dad30$@botecomm.com> <8D88662E-BB53-49F2-B003-2C9288A3FD03@jerris.com> Message-ID: <4D490B11-E40A-4598-B480-F203F1DAAF46@mgtech.com> FYI in case this affects Linux/Windows: removal of libvpx prereq install requires nasm or yasm to be added as prerequisite under OS X 10.9/10/11 since the OS X nasm is too old. Previous hombrew libvpx auto installed yasm. > On Feb 25, 2016, at 10:49 AM, Michael Jerris wrote: > > we are working on how exactly this will effect packaging and install instructions, but we will no longer have yuv or vpx pre-reqs and the requirement to rebuild libav against our vpx is also removed. > >> On Feb 25, 2016, at 1:39 PM, Bote Man wrote: >> >> Mike, I put a note to this effect on the Debian 8 installation page. >> >> I'm glad I caught this as I have been on a customer's site a lot and have >> not been able to keep with the -users mailing list. Lemme know if this >> changes so I can update it. >> >> Thanks. >> >> Bote >> >> >>> -----Original Message----- >>> From: Mario G >>> Sent: Thursday, 25 February, 2016 13:16 >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] major build change in master. >>> >>> This fixed OS X VPX issue, aok now. Questions: >>> >>> I assume this means YUV and VPX can be removed as prerequisites? >>> >>> Will this be one for 1.6 as well? >>> >>> Just asking so I can update the wiki and Applescript installer as needed. >>> Thanks! >>> >>> >>>> On Feb 24, 2016, at 3:49 PM, Michael Jerris wrote: >>>> >>>> please note this important change that went into master today. As of >>> today, we are no longer using system versions of libyuv and libvpx due to >>> major conflicts with system versions of these libraries. These are now >> built >>> static into the freeswitch core. Also note, mod_vpx no longer exists, it >> is >>> automatically loaded as part of the core and you will no longer have >>> mod_vpx.so or have to manually load it. I'll have more details coming, >> but let >>> me know if you have any questions. Please note for anyone doing cross >>> compiling, this probably needs a bit more work for you, I'll be working on >>> fixing that this week. >>>> >>>> Mike >>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 freeswitch at digitaldescent.net Sat Feb 27 01:45:03 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Fri, 26 Feb 2016 14:45:03 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> Message-ID: I think that solved it, thanks! I did have ACL configured, just apparently incorrectly. Out of the (fusionpbx) box I had two ACL lists, one called LAN with a default of allow, which I had entered my CIDR into. One called domain, which was default deny, but I had made an allow entry for my domain, but no CIDR. When I added my CIDR to the domain list, it started working. Thanks again for walking me through the troubleshooting, sorry for such a noob mistake :) On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein wrote: > Can you show a log of a internal call? > > Are the phones showing registered? Show registrations > > Did you by any chance add your ip range to the ACL. > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Thursday, February 25, 2016 8:18 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Moishe, > > Thanks for double checking this, I hadn't thought to try an extension > <-> extension call as it's a private server just for me, however, after > creating another extension, I cannot make internal calls. It seems, I can > only make outbound calls. > > > > Internal -> External is working. > > External -> Internal only works to the FS server, not to the extension. > > Internal -> Internal does not work. > > > > > > > > On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein > wrote: > > Does internal call also go direct to voicemail? > > Are you sure you don?t have a *99 before the extension number? > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Thursday, February 25, 2016 11:00 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Ok, in that case, what do I do next to figure out why inbound calls go > directly to voicemail instead of ringing the extension? > > On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: > > Ok, in that case, what do I look at next to see why it goes straight to > voicemail instead of ringing my extension? > > On Feb 25, 2016 5:52 AM, "Brian West" wrote: > > Its just the channel name so don't get tripped up by that, its just named > with some default data about the session. > > > > On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow < > freeswitch at digitaldescent.net> wrote: > > Debug console output: https://pastebin.freeswitch.org/24572 > > > > My fqdn only resolves inside the network, but the external-ip is > configured in FS to use stun, which reports my external ip. > > > > >Is your endpoint registered? How often is it set to reregister? > > Yes, and the zoiper default registration expiry is 3600 > > > > >Does your router have a sip alg? Are the ports opened in your firewall? > > > ish, no GUI for it, but I did make sure to disable it from the command > line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but > allow bi-directional with existing states. I do get 2 way audio when > calling out, and I also can hear the system forwarding my call in to > voicemail; the voicemail message and prompts. I know asterisk is different > from FS, but my asterisk experience has proven that at least with asterisk, > port forwarding isn't needed. Also, unfortunately, I cannot get a static > ip. > > > > Thanks! > > -Brian > > > > On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: > > It is probably sending call to your extension at domainname, if your > external ip is your domain name then you will see the external ip. > > Is your endpoint registered? How often is it set to reregister? > > Does your router have a sip alg? Are the ports opened in your firewall? > > Very hard to guess without a log of a call with debug enabled. > > sofia global sip trace on > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Wednesday, February 24, 2016 10:00 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Hello all! > > I'm new to freeswitch, so I'm sure this is just a newbie configuration > error. Sorry if it's been answered a million times, my google searches > always just bring up the standard NAT configuration pages. I've already > followed the confluence page on configuring NAT. > > > > My Setup: > > Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 > instance. > > SIP Provider: Flowroute > > > > NAT: extension and freeswitch are on the same network, both of which are > behind NAT and connecting to flowroute. I configured external sip and rtp > to use stun entries. External profile is using $${external_rtp/sip_ip} for > ext-rtp/sip respectively. > > > > I have one DID configured to go directly to my one extension. My > extension can register just fine. My extension can dial out. When I call > my cell, the call connects and I have 2 way audio. When I dial my DID from > my cell, I can see the call hitting the FS server, but instead of ringing > my extension, it goes straight to my extensions voicemail (which I can just > fine). > > > > When I look the at the console, it appears (sorry if this is wrong, I'm > only a day into free switch) that FS is attempting to route the call to my > extension...@ my external ip? > > > > I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] > switch_channel.c:1101 New Channel sofia/internal/@ > **:46072 > > > > > > Shouldn't that read sofia/internal/@ ? > > > > Sofia Status says: > > > > Name Type > Data State > > > ================================================================================================= > > external-ipv6 profile > sip:mod_sofia@[::1]:5080 RUNNING (0) > > external profile > sip:mod_sofia@:5080 RUNNING (0) > > external:: gateway sip:@ > sip.flowroute.com REGED > > internal-ipv6 profile > sip:mod_sofia@[::1]:5060 RUNNING (0) > > internal profile > sip:mod_sofia@:5060 RUNNING (0) > > > ================================================================================================= > > > > If I'm completely off base here, can anyone recommend where I can start > looking to change/troubleshoot the issue? I feel like it's just me missing > something, I just can't determine what that might be. > > > > Thanks, > > -Brian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160226/ddc44e74/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160226/ddc44e74/attachment-0001.jpg From aqsyounas at gmail.com Sat Feb 27 07:17:17 2016 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 27 Feb 2016 09:17:17 +0500 Subject: [Freeswitch-users] discriminate between session answered by users and voice mail automatic messages In-Reply-To: References: Message-ID: Or listen for answer events and apply avmd on outbound leg. On 26-Feb-2016 7:03 am, "?talo Rossi" wrote: > https://freeswitch.org/confluence/display/FREESWITCH/mod_com_amd > > On Thu, Feb 25, 2016 at 7:24 PM, sharigo roma > wrote: > >> Dear freeswitchers, >> >> is there a way to understand if a new session originated from a python >> handler has been actually answered by the remote user or by the operator >> voice mail with a message like "the customer you are trying to reach is not >> available at the moment" ? >> >> The python code looks something like the following: >> >> >> def handler(session, args): >> ## I do various things.... and than >> >> session_string = str("sofia/gateway/MY_GW/0039123456789?) >> new_session = freeswitch.Session(session_string) >> if new_session.ready(): >> # I get here also in case of automatic voice mail message >> else: >> # I supposed i?d get here in case of automatic ?unreachable? messages? >> >> >> Thanks in advance >> Lorenzo >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160227/ad9743e2/attachment.html From ynasida at gmail.com Sat Feb 27 11:36:46 2016 From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=) Date: Sat, 27 Feb 2016 11:36:46 +0300 Subject: [Freeswitch-users] where mod_com_g729 ? Message-ID: Hi list I have installed FS 1.6.6 via apt-get method (all modules) and don't see mod_com_g729 anywhere. Should I install it from source ? Please advice. # apt-cache search freeswitch | grep 729 freeswitch-mod-g729 - mod_g729 for FreeSWITCH freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160227/b9c8e182/attachment.html From shabbirabbasi92 at gmail.com Sat Feb 27 11:43:15 2016 From: shabbirabbasi92 at gmail.com (Shabbir abbasi) Date: Sat, 27 Feb 2016 13:43:15 +0500 Subject: [Freeswitch-users] Exception: ReferenceError: File is not defined (near: "var fd = new File(logfile); Message-ID: i am getting this error in javascript here is my code logToFile.js var logfile = "/usr/local/freeswitch/scripts/test.log"; var fd = new File(logfile); i have defined but error is coming logToFile.js:3 Exception: ReferenceError: File is not defined (near: "var fd = new File(logfile);") Dont know why ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160227/2116144b/attachment.html From gregor at infomedia.si Sat Feb 27 18:34:01 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 27 Feb 2016 16:34:01 +0100 Subject: [Freeswitch-users] originate and valet_park Message-ID: Is it possible to originate call and after answer park it in valet? I tried something like: originate user/1000 &valet_park mylot 5555 Extension rings and after pickup call is disconnected. Please for hint how to correctly format command or how to achive such scenario in different way. Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160227/4be194d7/attachment.html From italo at freeswitch.org Sat Feb 27 18:49:46 2016 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Sat, 27 Feb 2016 12:49:46 -0300 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? wrote: > Hi list > > I have installed FS 1.6.6 via apt-get method (all modules) and don't see > mod_com_g729 anywhere. > > Should I install it from source ? > > Please advice. > > # apt-cache search freeswitch | grep 729 > freeswitch-mod-g729 - mod_g729 for FreeSWITCH > freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160227/232d282c/attachment.html From piotrek.gregor at gmail.com Sat Feb 27 15:35:33 2016 From: piotrek.gregor at gmail.com (Peter Piotr) Date: Sat, 27 Feb 2016 12:35:33 +0000 Subject: [Freeswitch-users] discriminate between session answered by users and voice mail automatic messages In-Reply-To: References: Message-ID: Module avmd will not help much in this case as this can be used to detect sinusoidal tone of single frequency only (e.g. beep). Piotr On 27 February 2016 at 04:17, Aqs Younas wrote: > Or listen for answer events and apply avmd on outbound leg. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160227/8f3a2cbb/attachment-0001.html From avi at avimarcus.net Sat Feb 27 20:28:00 2016 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 27 Feb 2016 17:28:00 +0000 Subject: [Freeswitch-users] Exception: ReferenceError: File is not defined (near: "var fd = new File(logfile); In-Reply-To: References: Message-ID: <0100015323c5d1c5-4b44a886-5859-4e6b-8caa-f29583a6a8f4-000000@email.amazonses.com> The wiki seems to say that File isn't yet implemented for mod_v8... -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel) 020-3298-2875 (UK) On Sat, Feb 27, 2016 at 10:43 AM, Shabbir abbasi wrote: > i am getting this error in javascript here is my code > logToFile.js > > var logfile = "/usr/local/freeswitch/scripts/test.log"; > var fd = new File(logfile); > > i have defined but error is coming > logToFile.js:3 Exception: ReferenceError: File is not defined (near: "var > fd = new File(logfile);") > > Dont know why ? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160227/2eb9e341/attachment.html From ynasida at gmail.com Sat Feb 27 21:57:30 2016 From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=) Date: Sat, 27 Feb 2016 21:57:30 +0300 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: Looks like you didn't read my question. 2016-02-27 18:49 GMT+03:00 ?talo Rossi : > https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 > > On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? wrote: > >> Hi list >> >> I have installed FS 1.6.6 via apt-get method (all modules) and don't see >> mod_com_g729 anywhere. >> >> Should I install it from source ? >> >> Please advice. >> >> # apt-cache search freeswitch | grep 729 >> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160227/fb6445d9/attachment.html From brian at freeswitch.org Sat Feb 27 22:11:04 2016 From: brian at freeswitch.org (Brian West) Date: Sat, 27 Feb 2016 13:11:04 -0600 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: No he did read your question, there is no source to compile, did you read the page ?talo sent? On Saturday, February 27, 2016, ???? ?????? wrote: > Looks like you didn't read my question. > > 2016-02-27 18:49 GMT+03:00 ?talo Rossi >: > >> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 >> >> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? > > wrote: >> >>> Hi list >>> >>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see >>> mod_com_g729 anywhere. >>> >>> Should I install it from source ? >>> >>> Please advice. >>> >>> # apt-cache search freeswitch | grep 729 >>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 >> > > -- *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/20160227/e61e2967/attachment.html From krice at freeswitch.org Sun Feb 28 00:00:10 2016 From: krice at freeswitch.org (Ken Rice) Date: Sat, 27 Feb 2016 15:00:10 -0600 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: I'm pretty sure the link he gave you is the g729 commercial mod installation instructions and answers your question Sent from my iPhone > On Feb 27, 2016, at 12:57 PM, ???? ?????? wrote: > > Looks like you didn't read my question. > > 2016-02-27 18:49 GMT+03:00 ?talo Rossi : >> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 >> >>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? wrote: >>> Hi list >>> >>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see mod_com_g729 anywhere. >>> >>> Should I install it from source ? >>> >>> Please advice. >>> >>> # apt-cache search freeswitch | grep 729 >>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160227/c68cbc8e/attachment-0001.html From covici at ccs.covici.com Sun Feb 28 00:54:40 2016 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sat, 27 Feb 2016 16:54:40 -0500 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: <22770.1456610080@ccs.covici.com> Why use options ping at all -- I have never had good success with it, if it fails then it causes lots of problems, more than its supposed advantages. Seeliger, Adam wrote: > Hi, here is the requested output. > > I changed the real user, domain and ip address values into descriptions. > I guess the values are not necessary? > > Both, FreeSWITCH and the phone are in the same network (no NAT involved here) > > Registrations: > ================================================================================================= > Call-ID: 3134353634383833373032323631-ncwgvit2obfp > User: user at domain > Contact: "User Name" > Agent: snom715/8.7.5.35 > Status: Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912) > Ping-Status: Reachable > Host: hostname > IP: ip > Port: 60206 > Auth-User: user > Auth-Realm: domain > MWI-Account: user at domain > > Total items returned: 1 > ================================================================================================= > 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable > 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure > > As you can see, the user immediately got unregistered again ? > > Best regards, > Adam > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov > Gesendet: Freitag, 26. Februar 2016 15:45 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue > > Please send output of command "sofia status profile internal reg " > > On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam > wrote: > Hi, > > the phone uses sip+tls. > I test using a snom715, got plenty other phones here, but I guess they will behave the same way. > It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls) > > Regards > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov > Gesendet: Freitag, 26. Februar 2016 13:59 > > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue > > If your phone has enabled SIPS uri please disable and use sip+tls. > > > On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: > Hi and thanks for the feedback, > > sry that I did not respond for a long time. > > I already use: > > > I also tested all mentioned params below, nothing works. > > When I register a User via TLS FreeSWITCH does not even try to ping the user. > I turned sofia global siptrace on and watched the flow: > > User Server > 13:09:33.311446: REGISTER [TLS] -> > 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] > 13:09:33.331948: REGISTER (AUTH) [TLS] -> > 13:09:33.336959: <- 200 OK [TLS] > Nothing happens > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable > 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure > > When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered. > > Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration? > > Thanks in advance, > Adam > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah > Gesendet: Freitag, 29. Januar 2016 09:25 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue > > Hi! > This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS. > I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set. > I found out that in my case, what works best even with TLS, is to use: > > This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports. > Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead. > > You could also look into: > > --> > --> > > > > I?ll leave it up to you to investigate those options more in details on the FS documentation. > > Please keep us posted! > > E > On Jan 28, 2016, at 11:48 AM, Seeliger, Adam > wrote: > > Hi all, > > I have a problem, when I enable TLS and register a phone using TLS on Port 5061. > FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone. > Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?. > > I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS. > Is there any way to ping TLS registered using TLS? > > Thanks in advance > > - Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 max at nysolutions.com Sun Feb 28 03:58:55 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 28 Feb 2016 00:58:55 +0000 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> Message-ID: You should not need to configure an acl for your cidr unless your endpoints are not registering. The only acl you would need is for your sip providers ip?s if they are ip authenticating and sending traffic to your internal profile. 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 Brian Chow Sent: Friday, February 26, 2016 5:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? I think that solved it, thanks! I did have ACL configured, just apparently incorrectly. Out of the (fusionpbx) box I had two ACL lists, one called LAN with a default of allow, which I had entered my CIDR into. One called domain, which was default deny, but I had made an allow entry for my domain, but no CIDR. When I added my CIDR to the domain list, it started working. Thanks again for walking me through the troubleshooting, sorry for such a noob mistake :) On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein > wrote: Can you show a log of a internal call? Are the phones showing registered? Show registrations Did you by any chance add your ip range to the ACL. 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 Brian Chow Sent: Thursday, February 25, 2016 8:18 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Moishe, Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls. It seems, I can only make outbound calls. Internal -> External is working. External -> Internal only works to the FS server, not to the extension. Internal -> Internal does not work. On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein > wrote: Does internal call also go direct to voicemail? Are you sure you don?t have a *99 before the extension number? 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 Brian Chow Sent: Thursday, February 25, 2016 11:00 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension? On Feb 25, 2016 5:52 AM, "Brian West" > wrote: Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow > wrote: Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip. >Is your endpoint registered? How often is it set to reregister? Yes, and the zoiper default registration expiry is 3600 >Does your router have a sip alg? Are the ports opened in your firewall? > ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip. Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on 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 Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@:46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20160228/70116943/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/20160228/70116943/attachment-0001.jpg From freeswitch at digitaldescent.net Sun Feb 28 06:03:41 2016 From: freeswitch at digitaldescent.net (Brian Chow) Date: Sat, 27 Feb 2016 19:03:41 -0800 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> Message-ID: The extensions were always registering. The clients reported as registered, and I was able to make outbound calls without issue. It was only inbound calls that went directly to voicemail instead of ringing. I also could not call extension to extension. After adding the cidr to the acl list both internal extension calls and inbound calls were working. However, it sounds like you believe that should not be the case based on that change? I can post config samples later when I get back home. Or at least fusionpbx screenshots. The I'm not sure where fusionpbx gets all the settings from. The internal and external profiles all have .noload extensions, which seems different from the other FreeSWITCH config files. On Feb 27, 2016 5:01 PM, "Moishe Grunstein" wrote: > You should not need to configure an acl for your cidr unless your > endpoints are not registering. > > The only acl you would need is for your sip providers ip?s if they are ip > authenticating and sending traffic to your internal profile. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Friday, February 26, 2016 5:45 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > I think that solved it, thanks! I did have ACL configured, just > apparently incorrectly. Out of the (fusionpbx) box I had two ACL lists, > one called LAN with a default of allow, which I had entered my CIDR into. > One called domain, which was default deny, but I had made an allow entry > for my domain, but no CIDR. When I added my CIDR to the domain list, it > started working. > > > > Thanks again for walking me through the troubleshooting, sorry for such a > noob mistake :) > > > > On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein > wrote: > > Can you show a log of a internal call? > > Are the phones showing registered? Show registrations > > Did you by any chance add your ip range to the ACL. > > > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Thursday, February 25, 2016 8:18 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Moishe, > > Thanks for double checking this, I hadn't thought to try an extension > <-> extension call as it's a private server just for me, however, after > creating another extension, I cannot make internal calls. It seems, I can > only make outbound calls. > > > > Internal -> External is working. > > External -> Internal only works to the FS server, not to the extension. > > Internal -> Internal does not work. > > > > > > > > On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein > wrote: > > Does internal call also go direct to voicemail? > > Are you sure you don?t have a *99 before the extension number? > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Thursday, February 25, 2016 11:00 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Ok, in that case, what do I do next to figure out why inbound calls go > directly to voicemail instead of ringing the extension? > > On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: > > Ok, in that case, what do I look at next to see why it goes straight to > voicemail instead of ringing my extension? > > On Feb 25, 2016 5:52 AM, "Brian West" wrote: > > Its just the channel name so don't get tripped up by that, its just named > with some default data about the session. > > > > On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow < > freeswitch at digitaldescent.net> wrote: > > Debug console output: https://pastebin.freeswitch.org/24572 > > > > My fqdn only resolves inside the network, but the external-ip is > configured in FS to use stun, which reports my external ip. > > > > >Is your endpoint registered? How often is it set to reregister? > > Yes, and the zoiper default registration expiry is 3600 > > > > >Does your router have a sip alg? Are the ports opened in your firewall? > > > ish, no GUI for it, but I did make sure to disable it from the command > line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but > allow bi-directional with existing states. I do get 2 way audio when > calling out, and I also can hear the system forwarding my call in to > voicemail; the voicemail message and prompts. I know asterisk is different > from FS, but my asterisk experience has proven that at least with asterisk, > port forwarding isn't needed. Also, unfortunately, I cannot get a static > ip. > > > > Thanks! > > -Brian > > > > On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: > > It is probably sending call to your extension at domainname, if your > external ip is your domain name then you will see the external ip. > > Is your endpoint registered? How often is it set to reregister? > > Does your router have a sip alg? Are the ports opened in your firewall? > > Very hard to guess without a log of a call with debug enabled. > > sofia global sip trace on > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: 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 *Brian Chow > *Sent:* Wednesday, February 24, 2016 10:00 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to > internal extension with external IP? > > > > Hello all! > > I'm new to freeswitch, so I'm sure this is just a newbie configuration > error. Sorry if it's been answered a million times, my google searches > always just bring up the standard NAT configuration pages. I've already > followed the confluence page on configuring NAT. > > > > My Setup: > > Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 > instance. > > SIP Provider: Flowroute > > > > NAT: extension and freeswitch are on the same network, both of which are > behind NAT and connecting to flowroute. I configured external sip and rtp > to use stun entries. External profile is using $${external_rtp/sip_ip} for > ext-rtp/sip respectively. > > > > I have one DID configured to go directly to my one extension. My > extension can register just fine. My extension can dial out. When I call > my cell, the call connects and I have 2 way audio. When I dial my DID from > my cell, I can see the call hitting the FS server, but instead of ringing > my extension, it goes straight to my extensions voicemail (which I can just > fine). > > > > When I look the at the console, it appears (sorry if this is wrong, I'm > only a day into free switch) that FS is attempting to route the call to my > extension...@ my external ip? > > > > I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] > switch_channel.c:1101 New Channel sofia/internal/@ > **:46072 > > > > > > Shouldn't that read sofia/internal/@ ? > > > > Sofia Status says: > > > > Name Type > Data State > > > ================================================================================================= > > external-ipv6 profile > sip:mod_sofia@[::1]:5080 RUNNING (0) > > external profile > sip:mod_sofia@:5080 RUNNING (0) > > external:: gateway sip:@ > sip.flowroute.com REGED > > internal-ipv6 profile > sip:mod_sofia@[::1]:5060 RUNNING (0) > > internal profile > sip:mod_sofia@:5060 RUNNING (0) > > > ================================================================================================= > > > > If I'm completely off base here, can anyone recommend where I can start > looking to change/troubleshoot the issue? I feel like it's just me missing > something, I just can't determine what that might be. > > > > Thanks, > > -Brian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160227/81977af3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160227/81977af3/attachment-0001.jpg From max at nysolutions.com Sun Feb 28 06:47:48 2016 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 28 Feb 2016 03:47:48 +0000 Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? In-Reply-To: References: <23b1626b3814448b90842a279dc30be5@nysolutions.com> <62e0a3e6c4e14d40b15aedb1d3e1b3b6@nysolutions.com> Message-ID: Your problem is what you initially added to the ACL, if you remove it all and reload ACL, it should work fine. 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 Brian Chow Sent: Saturday, February 27, 2016 10:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? The extensions were always registering. The clients reported as registered, and I was able to make outbound calls without issue. It was only inbound calls that went directly to voicemail instead of ringing. I also could not call extension to extension. After adding the cidr to the acl list both internal extension calls and inbound calls were working. However, it sounds like you believe that should not be the case based on that change? I can post config samples later when I get back home. Or at least fusionpbx screenshots. The I'm not sure where fusionpbx gets all the settings from. The internal and external profiles all have .noload extensions, which seems different from the other FreeSWITCH config files. On Feb 27, 2016 5:01 PM, "Moishe Grunstein" > wrote: You should not need to configure an acl for your cidr unless your endpoints are not registering. The only acl you would need is for your sip providers ip?s if they are ip authenticating and sending traffic to your internal profile. 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 Brian Chow Sent: Friday, February 26, 2016 5:45 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? I think that solved it, thanks! I did have ACL configured, just apparently incorrectly. Out of the (fusionpbx) box I had two ACL lists, one called LAN with a default of allow, which I had entered my CIDR into. One called domain, which was default deny, but I had made an allow entry for my domain, but no CIDR. When I added my CIDR to the domain list, it started working. Thanks again for walking me through the troubleshooting, sorry for such a noob mistake :) On Thu, Feb 25, 2016 at 5:50 PM, Moishe Grunstein > wrote: Can you show a log of a internal call? Are the phones showing registered? Show registrations Did you by any chance add your ip range to the ACL. 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 Brian Chow Sent: Thursday, February 25, 2016 8:18 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Moishe, Thanks for double checking this, I hadn't thought to try an extension <-> extension call as it's a private server just for me, however, after creating another extension, I cannot make internal calls. It seems, I can only make outbound calls. Internal -> External is working. External -> Internal only works to the FS server, not to the extension. Internal -> Internal does not work. On Thu, Feb 25, 2016 at 8:12 AM, Moishe Grunstein > wrote: Does internal call also go direct to voicemail? Are you sure you don?t have a *99 before the extension number? 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 Brian Chow Sent: Thursday, February 25, 2016 11:00 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" > wrote: Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension? On Feb 25, 2016 5:52 AM, "Brian West" > wrote: Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow > wrote: Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip. >Is your endpoint registered? How often is it set to reregister? Yes, and the zoiper default registration expiry is 3600 >Does your router have a sip alg? Are the ports opened in your firewall? > ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip. Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein > wrote: It is probably sending call to your extension at domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on 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 Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/@:46072 Shouldn't that read sofia/internal/@ ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@:5080 RUNNING (0) external:: gateway sip:@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160228/b527346b/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/20160228/b527346b/attachment-0001.jpg From mgiammarco at gmail.com Sun Feb 28 14:00:31 2016 From: mgiammarco at gmail.com (Mario Giammarco) Date: Sun, 28 Feb 2016 11:00:31 +0000 (UTC) Subject: [Freeswitch-users] Recording and mono directional webrtc video References: Message-ID: Mario Giammarco writes: > I need a bidirectional audio > call with one direction video only from caller. And I need to record audio > and video from caller. Hello, nobody can help me? I am asking only if it is possible and supported. Thanks, Mario From shlomis at liveperson.com Sun Feb 28 18:17:55 2016 From: shlomis at liveperson.com (Shlomi Schwartz) Date: Sun, 28 Feb 2016 17:17:55 +0200 Subject: [Freeswitch-users] MOD_VERTO Documentation Message-ID: Hi all, Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any documentation regarding the dialogParams on verto.invite method? The only documentation I found was : https://freeswitch.org/confluence/display/FREESWITCH/mod_verto Thanks :) -- Shlomi Schwartz R&D Technical Leader T: +972-74-700-4511 We Create Meaningful Connections -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160228/64c4a7a7/attachment.html From italo at freeswitch.org Sun Feb 28 20:14:17 2016 From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=) Date: Sun, 28 Feb 2016 09:14:17 -0800 (PST) Subject: [Freeswitch-users] MOD_VERTO Documentation In-Reply-To: References: Message-ID: Hi Shlomi, We have a detailed doc here, but using the javascript lib, maybe it can give you some light, take a look at http://evoluxbr.github.io/verto-docs/ > On Feb 28 2016, at 1:08 pm, Shlomi Schwartz <shlomis at liveperson.com> wrote: > > Hi all, > > > > Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any documentation regarding the dialogParams on verto.invite method? > > > > The only documentation I found was : > > > > Thanks :) > > > > \-- > > ![](https://signature.s3.amazonaws.com/2015/lp_logo.png) > --- > Shlomi Schwartz > R&D Technical Leader > T: +972-74-700-4511 > | | [![](https://signature.s3.amazonaws.com/2015/LinkedIn.png)](http://www. linkedin.com/company/164748) | [![](https://signature.s3.amazonaws.com/2015/Tw itter.png)](http://twitter.com/liveperson) | [![](https://signature.s3.amazona ws.com/2015/Facebook.png)](http://www.facebook.com/LivePersonInc) > ---|---|--- > We Create Meaningful Connections > [![](https://signature.s3.amazonaws.com/2015/banners/Inc_email_banner_2.8.20 16.jpg)](http://info.liveperson.com/201602/liz-welch/liveperson-robert- locascio-on-death-of-the-800-call.html) > > > > This message may contain confidential and/or privileged information. > > If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. > > If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160228/e2b9cc23/attachment.html From lists at kavun.ch Sun Feb 28 21:33:49 2016 From: lists at kavun.ch (Emrah) Date: Sun, 28 Feb 2016 19:33:49 +0100 Subject: [Freeswitch-users] SRTP breaks my TLS session In-Reply-To: <93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch> References: <84F64031-B943-4671-A0BC-66FC02DF7C51@kavun.ch> <88ACC7C7-589C-4CC8-A146-3DBF31B84A2E@mgtech.com> <93C84919-0D70-4DD3-BB86-46DE87492F9C@kavun.ch> Message-ID: <14EF2D3F-9034-4D59-BC37-16B6E20BAEE8@kavun.ch> Hello there, I can confirm that a PCAP gives me a bunch of TLS packets. How do you suggest debugging this? Thanks! > On Feb 26, 2016, at 10:36 AM, Emrah wrote: > > Thanks for this. > > This isn?t just a yealink thing. I?ve encountered sporadic issues with soft phones and other desk phones as well. > > I didn?t use the PCap capture feature because I had assumed it would give me a bunch of TLS packets. I?ll test that and revert back. > > How can we explain that I have more calls failing if I register multiple accounts? > > Emrah > >> On Feb 26, 2016, at 6:37 AM, Mario G > wrote: >> >> You may want to run a pcap trace on the Yealink. It?s under settings->Configuration. Start/test/export. >> >>> On Feb 25, 2016, at 3:35 PM, Brian West > wrote: >>> >>> Sounds like there may be a bug in the yealink if packet size is of issue on TCP/TLS connections. >>> >>> On Thu, Feb 25, 2016 at 5:13 PM, Emrah > wrote: >>> Hello list, >>> I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size. >>> Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS. >>> Incoming calls work reliably with or without SRTP. >>> >>> How do you suggest debugging this? >>> I tried setting up a fresh instance of FS but the issue persists. >>> Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device. >>> >>> Any suggestion is welcome. Have you experienced this? >>> >>> I?m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95. >>> >>> 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 >>> >>> >>> >>> -- >>> 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/20160228/ec4931d6/attachment-0001.html From gregor at infomedia.si Sun Feb 28 21:37:45 2016 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 28 Feb 2016 19:37:45 +0100 Subject: [Freeswitch-users] MOD_VERTO Documentation In-Reply-To: References: Message-ID: This tutorial is great! Thank you. 2016-02-28 18:14 GMT+01:00 ?talo Rossi : > Hi Shlomi, > > We have a detailed doc here, but using the javascript lib, maybe it can > give you some light, take a look at http://evoluxbr.github.io/verto-docs/ > >> On Feb 28 2016, at 1:08 pm, Shlomi Schwartz >> wrote: >> Hi all, >> >> Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any >> documentation regarding the dialogParams on verto.invite method? >> >> The only documentation I found was : >> https://freeswitch.org/confluence/display/FREESWITCH/mod_verto >> >> Thanks :) >> >> -- >> Shlomi Schwartz >> R&D Technical Leader >> T: +972-74-700-4511 >> >> We Create Meaningful Connections >> >> >> >> This message may contain confidential and/or privileged information. >> If you are not the addressee or authorized to receive this on behalf of >> the addressee you must not use, copy, disclose or take action based on this >> message or any information herein. >> If you have received this message in error, please advise the sender >> immediately by reply email and delete this message. 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/20160228/74805749/attachment.html From ssinyagin at gmail.com Mon Feb 29 00:20:20 2016 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sun, 28 Feb 2016 22:20:20 +0100 Subject: [Freeswitch-users] Recording and mono directional webrtc video In-Reply-To: References: Message-ID: Doesn't seem impossible, but needs testing. If you have a budget, there are some people who can help you build a prototype and troubleshoot the problems if there are any. On 28 Feb 2016 17:06, "Mario Giammarco" wrote: > Mario Giammarco writes: > > > I need a bidirectional audio > > call with one direction video only from caller. And I need to record > audio > > and video from caller. > > Hello, > nobody can help me? I am asking only if it is possible and supported. > > Thanks, > Mario > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160228/81795b18/attachment.html From freeswitch.opencode at spamgourmet.com Mon Feb 29 01:43:05 2016 From: freeswitch.opencode at spamgourmet.com (freeswitch.opencode at spamgourmet.com) Date: Sun, 28 Feb 2016 22:43:05 +0000 Subject: [Freeswitch-users] Getting 503 when starting automatically In-Reply-To: References: , Message-ID: I enabled sofia tracing while the problem was occurring. I haven't had a chance to look into the source code yet, but so far I'm guessing it's not a permissions issue and something more like the invalid connection handle caching hypothesis. Here's the sofia log snippet (DNS names and IP addresses sanitized): nta.c:10803 outgoing_query_a() nta: for "myvoip.example.com" query "myvoip.example.com" A (cached) nta.c:10856 outgoing_answer_a() nta: myvoip.example.com. IN A 10.0.0.1 tport.c:3257 tport_tsend() tport_tsend(0xb5b392e8) tpn = udp/10.0.0.1:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.0.0.1:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb5b392e8): not found by name udp/10.0.0.1:5060 tport.c:3636 tport_send_fatal() tport_vsend(0xb5b392e8): Invalid argument with (s=22 udp/10.0.0.1:5060) tport.c:3492 tport_send_msg() tport_vsend returned -1 David From vishal.sharma at knowlarity.com Mon Feb 29 07:54:13 2016 From: vishal.sharma at knowlarity.com (Vishal Sharma) Date: Mon, 29 Feb 2016 10:24:13 +0530 Subject: [Freeswitch-users] File not found Message-ID: Hi, I am using FS 1.6.5, when FS tries to play a file (/doesnotexist/srv/sounds/automation/media-sec.mp3) which doesn't exist on file system, it takes 20 seconds for it to respond that file is not present and dead air is played during this time. On FS 1.4 and 1.2, it used to respond instantly. Am i missing some setting ... Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/333270a4/attachment.html From vishal.sharma at knowlarity.com Mon Feb 29 08:50:59 2016 From: vishal.sharma at knowlarity.com (Vishal Sharma) Date: Mon, 29 Feb 2016 11:20:59 +0530 Subject: [Freeswitch-users] Channel Hangup not received for Leg A Message-ID: I used following call string to bridge legB to legA {ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222} [leg_timeout=30]freetdm/outgoing/r/09506360596:_:{ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222} [leg_timeout=30]freetdm/outgoing/r/08601224488 If legA disconnect before any of LegB is bridged, I get Cannel_hangup for both legBs, and channel_hangup_complete for legA but no channel_hangup for legA. I am using 1.6.5 Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/5b56dbf2/attachment-0001.html From andrew.keil at visytel.com Mon Feb 29 09:37:02 2016 From: andrew.keil at visytel.com (Andrew Keil) Date: Mon, 29 Feb 2016 06:37:02 +0000 Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function In-Reply-To: References: Message-ID: Thanks for your response. I have gone through these with no luck. Like I said the session:destroy(???) crashes FreeSWITCH, which is therefore off the list. The rest simply interrupt the current function and do no end the script. I guess my next move is to see why session:destroy() crashes FreeSWITCH, however I am a little snowed under at the moment so if anyone has some time to replicate this (only needs one line of code in a Lua script) and pass this on to the developers that would be great. Andrew From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abaci B Sent: Saturday, 27 February 2016 1:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function See https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook for a few ways to exit the lua script (error(), return "exit", return "die", s:destroy("error message")). I personally tried return "exit" but it seems to me that it only exits the calling function, haven't had a chance to look further, it's possible that the calling it from the within a function is different. if you play around and figure out please report back. On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil > wrote: To FreeSWITCH Users, See below for a sample template for a Lua Service Script running inside FreeSWITCH. The issue I have is fairly straightforward. I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script. This function is CleanUp(). Then I would like the service to end. The problem I am having is if the caller hangs up during the playback of ?intro.wav? (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()). What actually happens is it returns to MainService() and continues to try and play ?info.wav?, unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call. My aim is to reduce extra code and to make the Lua script simpler and easier to read. Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc?. So is there a way to stop a Lua script running inside FreeSWITCH cleanly? I have tried the os.exit() this is barred from use by FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution. FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use ?if (session:ready()) then ?.? or ?if (not session:ready()) then goto HANGUPEXIT end? makes the code ugly. Thanks in advance, Andrew Keil Visytel Pty Ltd ------------------------------------------------------------------------------ Sample Lua Service ----------------------------------------------------------------------- -- Lua template for FreeSWITCH service -- By: Andrew Keil (Visytel Pty Ltd) -- Email: support at visytel.com -- Setup script wide variables here function PreAnswer() freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); -- Add your pre answer code from here -- End of your pre answer code freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n"); end function AnswerCaller() session:answer() session:sleep(1000) end function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n"); if (session:ready()) then -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end -- Add your main service code from here (caller would have been answered) session:streamFile("intro.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("info.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("outro.wav") if (not session:ready()) then goto HANGUPEXIT end -- End of your main service code end ::ENDSERVICE:: if (session:ready()) then -- End of service so hangup session:hangup() -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage end goto END ::HANGUPEXIT:: freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n"); ::END:: freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n"); end function CleanUp() freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); -- Add your cleanup code from here (caller would have been disconnected) -- End of your cleanup code freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n"); end function myHangupHook(s, status, arg) session:hangup() CleanUp() -- Run CleanUp function now since the caller has disconnected end -- Setup Hangup event handler here v_hangup = "HANGUP" session:setHangupHook("myHangupHook", "v_hangup") -- Call service functions in order PreAnswer() AnswerCaller() MainService() -- End of Lua service _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160229/faa69a31/attachment-0001.html From Adam.Seeliger at qsc.de Mon Feb 29 11:29:11 2016 From: Adam.Seeliger at qsc.de (Seeliger, Adam) Date: Mon, 29 Feb 2016 08:29:11 +0000 Subject: [Freeswitch-users] all-reg-options-ping and tls issue In-Reply-To: References: <2E67ADAE1D3582409C90EBAE8C64C5070149F6B3@QSCDEMXP01a.ONE4ALL.LAN> <6D3DEA47-DECB-4509-A327-B8DFEC1C1A83@kavun.ch> <2E67ADAE1D3582409C90EBAE8C64C507122D35C1@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D35E9@QSCDEMXP01a.ONE4ALL.LAN> <2E67ADAE1D3582409C90EBAE8C64C507122D361C@QSCDEMXP01a.ONE4ALL.LAN> Message-ID: <2E67ADAE1D3582409C90EBAE8C64C507122D372E@QSCDEMXP01a.ONE4ALL.LAN> Hi, I also tested it on latest master: FreeSWITCH Version 1.7.0+git~20160227T004333Z~d89a0ad52d~64bit (git d89a0ad 2016-02-27 00:43:33Z 64bit) FreeSWITCH still does not send OPTIONs to TLS registered Users and kills their Registration ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 16:53 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Registration is correct. Think it bug and requred to fill a jira tiket. I use master 4 mouth old and it work correctly. On Fri, Feb 26, 2016, 18:00 Seeliger, Adam > wrote: Hi, here is the requested output. I changed the real user, domain and ip address values into descriptions. I guess the values are not necessary? Both, FreeSWITCH and the phone are in the same network (no NAT involved here) Registrations: ================================================================================================= Call-ID: 3134353634383833373032323631-ncwgvit2obfp User: user at domain Contact: "User Name" Agent: snom715/8.7.5.35 Status: Registered(AUTO-NAT-2.0)(unknown) EXP(2016-02-26 16:21:32) EXPSECS(1912) Ping-Status: Reachable Host: hostname IP: ip Port: 60206 Auth-User: user Auth-Realm: domain MWI-Account: user at domain Total items returned: 1 ================================================================================================= 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5769 Sip user 'user at domain' is now Unreachable 2016-02-26 15:50:09.326648 [WARNING] sofia.c:5780 Expire sip user 'user at domain' due to options failure As you can see, the user immediately got unregistered again ? Best regards, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 15:45 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Please send output of command "sofia status profile internal reg " On Fri, Feb 26, 2016 at 4:21 PM, Seeliger, Adam > wrote: Hi, the phone uses sip+tls. I test using a snom715, got plenty other phones here, but I guess they will behave the same way. It really looks like FreeSWITCH is doing something wrong (or is wrongly configured ? if there are any parameters for options ping supporting both, udp and tls) Regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sergey Safarov Gesendet: Freitag, 26. Februar 2016 13:59 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue If your phone has enabled SIPS uri please disable and use sip+tls. On Fri, Feb 26, 2016 at 3:37 PM, Seeliger, Adam > wrote: Hi and thanks for the feedback, sry that I did not respond for a long time. I already use: I also tested all mentioned params below, nothing works. When I register a User via TLS FreeSWITCH does not even try to ping the user. I turned sofia global siptrace on and watched the flow: User Server 13:09:33.311446: REGISTER [TLS] -> 13:09:33.312552: <- 401 UNAUTHORIZED [TLS] 13:09:33.331948: REGISTER (AUTH) [TLS] -> 13:09:33.336959: <- 200 OK [TLS] Nothing happens 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5769 Sip user 'user at host' is now Unreachable 2016-02-26 13:10:00.619525 [WARNING] sofia.c:5780 Expire sip user 'user at host' due to options failure When I REGISTER the User via UDP FreeSWITCH starts to ping (OPTIONS) the user as soon as he is registered. Is there any way to force FreeSWITCH to send OPTIONs in both, udp and tls, depending on the registration? Thanks in advance, Adam Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Emrah Gesendet: Freitag, 29. Januar 2016 09:25 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] all-reg-options-ping and tls issue Hi! This is interesting. I experienced something rather similar where calls would drop because FS would timeout on certain packets sent over UDP instead of TLS. I assume you mean FS exits with port 5060 instead of port 5061? Because the port on the remote end should be dynamically set. I found out that in my case, what works best even with TLS, is to use: This goes as far as it can to lay out the path to contacting the client with all consideration in regards to NAT and dynamic ports. Not sure if it will help you. I?ve personally disabled options-ping an let my clients deal with keep-alive instead. You could also look into: --> --> I?ll leave it up to you to investigate those options more in details on the FS documentation. Please keep us posted! E On Jan 28, 2016, at 11:48 AM, Seeliger, Adam > wrote: Hi all, I have a problem, when I enable TLS and register a phone using TLS on Port 5061. FreeSWITCH still tries to ?ping? the phone using Port 5060 using UDP, which is ignored by the phone. Moments later FreeSWITCH deletes the registration, because ?unregister-on-options-fail? is set to ?true?. I already figured out, that you can set ?all-reg-options-ping? to ?udp-only?, but this would completely disable this feature for TLS. Is there any way to ping TLS registered using TLS? Thanks in advance - Adam _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160229/5430e10e/attachment-0001.html From dominique.jeannerod at interact-iv.com Mon Feb 29 11:52:12 2016 From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod) Date: Mon, 29 Feb 2016 09:52:12 +0100 Subject: [Freeswitch-users] mod_distributor / gwlist down In-Reply-To: References: Message-ID: Hello, I worked again on this, and there is no problem at all related with mod_distributor. What I found : there is a problem evaluating nested expressions. This code works as expected : This code is not OK : ? Any idea on the right syntax to make it work on one line as it should ? Best regards Dominique Jeannerod -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/f69ebdf4/attachment.html From idokan at gmail.com Mon Feb 29 15:59:47 2016 From: idokan at gmail.com (ik) Date: Mon, 29 Feb 2016 14:59:47 +0200 Subject: [Freeswitch-users] Debugging Perl code In-Reply-To: References: Message-ID: Hello, I have Perl code that does new freeswitch::Session. When that happens, freeswitch crashes. It's an old freeswitch v1.4.26. I execute it from now using the cli either by perl or perlrun. How can I debug the cause of such crash? Thank you Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/b40f82b1/attachment.html From rutu.patel at inextrix.com Mon Feb 29 16:15:28 2016 From: rutu.patel at inextrix.com (Rutu Patel) Date: Mon, 29 Feb 2016 18:45:28 +0530 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: References: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> Message-ID: Hi Jurijs, We have to consider the setup with freeswitch and opensips only. registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) And there is a firewall and we have added x.x.x.3 IP there. Is it possible there is some network related issue Or any sip profile parameter we need to set? -- Thanks, Rutu Patel On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga wrote: > Hi, > > 1) You have very complex set-up and I doubt that you need it. > > 2) As far as you have user with ip x.x.x.174 and opensips server with same > ip x.x.x.174 it very hard to debug. So I propose you to send new log where > will be difference between user ip and opensips IP. > > 3) If you have possibility, try to register directly with a user to > x.x.x.3 gateway and check if same issue still exists, if there is no such > issue anymore, then thee is definitely issue in your opensips x.x.x.174 and > freeswitch x.x.x.166. My point here is that you need to isolate issue and > to understand what part of your set-up works as expected and what is faulty. > > With kind regards, > > Jurijs > > 2016-02-23 7:51 GMT+02:00 Rutu Patel : > >> Thanks for the reply. >> >> Got your point about NATing issue and no response of 200 OK and as a >> resoult ACK Timeout. >> So, now to resolve the issue, if you can assist, what could be the >> possible fixies? >> From where can i start? where to look? >> >> Thanks. >> >> -- >> Thanks, >> Rutu Patel >> >> >> >> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: >> >>> 5 second response: 32 seconds is a timer/[network/NAT] issue. >>> >>> You have lots of 200s to the user since it's waiting for an ACK and >>> keeps retrying, but for whatever network reason (router... sip alg?), it >>> isn't getting one, so it triggers a timer to stop the call. >>> >>> >>> -Avi Marcus >>> BestFone >>> >>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >>> wrote: >>> >>>> Hello, >>>> >>>> Having issue of call dropping after 32 seconds, here are the details- >>>> >>>> x.x.x.174: opensips server >>>> x.x.x.166: freeswitch server >>>> x.x.x.3: another opensips server which is registered as gateway on >>>> above freeswitch server >>>> x.x.x.6: freeswitch server >>>> x.x.x.47: server through which the user is registered >>>> I am trying to call from xxxx9 to xxxxxxx29858 >>>> xxxxxxx00181 is caller-id name and caller-id number >>>> >>>> Call flow is like this: >>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips >>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>>> >>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>>> >>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>>> 2) Call hit the freeswitch server x.x.x.166 >>>> 3) After '180 Ringing' and '183 Session Progress' packet >>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>>> gateway x.x.x.3 >>>> But after 32 seconds call is dropped, >>>> Within 32 seconds audio is ok from both end so it should not be the RTP >>>> issue. >>>> Here I have attached the file with sip logs, you can observer from the >>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>>> >>>> What is wrong here? Any help would be appreciated here. >>>> >>>> Here is the file with sip logs >>>> -- >>>> Thanks, >>>> Rutu Patel >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >> > > > Jurijs > > 2016-02-23 7:51 GMT+02:00 Rutu Patel : > >> Thanks for the reply. >> >> Got your point about NATing issue and no response of 200 OK and as a >> resoult ACK Timeout. >> So, now to resolve the issue, if you can assist, what could be the >> possible fixies? >> From where can i start? where to look? >> >> Thanks. >> >> -- >> Thanks, >> Rutu Patel >> >> >> >> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: >> >>> 5 second response: 32 seconds is a timer/[network/NAT] issue. >>> >>> You have lots of 200s to the user since it's waiting for an ACK and >>> keeps retrying, but for whatever network reason (router... sip alg?), it >>> isn't getting one, so it triggers a timer to stop the call. >>> >>> >>> -Avi Marcus >>> BestFone >>> >>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >>> wrote: >>> >>>> Hello, >>>> >>>> Having issue of call dropping after 32 seconds, here are the details- >>>> >>>> x.x.x.174: opensips server >>>> x.x.x.166: freeswitch server >>>> x.x.x.3: another opensips server which is registered as gateway on >>>> above freeswitch server >>>> x.x.x.6: freeswitch server >>>> x.x.x.47: server through which the user is registered >>>> I am trying to call from xxxx9 to xxxxxxx29858 >>>> xxxxxxx00181 is caller-id name and caller-id number >>>> >>>> Call flow is like this: >>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips >>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>>> >>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>>> >>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>>> 2) Call hit the freeswitch server x.x.x.166 >>>> 3) After '180 Ringing' and '183 Session Progress' packet >>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>>> gateway x.x.x.3 >>>> But after 32 seconds call is dropped, >>>> Within 32 seconds audio is ok from both end so it should not be the RTP >>>> issue. >>>> Here I have attached the file with sip logs, you can observer from the >>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>>> >>>> What is wrong here? Any help would be appreciated here. >>>> >>>> Here is the file with sip logs >>>> -- >>>> Thanks, >>>> Rutu Patel >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20160229/a38f7d9f/attachment-0001.html From ynasida at gmail.com Mon Feb 29 16:20:31 2016 From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=) Date: Mon, 29 Feb 2016 16:20:31 +0300 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: Thanks guys. I just had to set manually the path for script fs-latest-installer-v1.6 Otherwise mod_com_g729.so was not installed anywhere (without any warning message). 2016-02-28 0:00 GMT+03:00 Ken Rice : > I'm pretty sure the link he gave you is the g729 commercial mod > installation instructions and answers your question > > Sent from my iPhone > > On Feb 27, 2016, at 12:57 PM, ???? ?????? wrote: > > Looks like you didn't read my question. > > 2016-02-27 18:49 GMT+03:00 ?talo Rossi : > >> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 >> >> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? wrote: >> >>> Hi list >>> >>> I have installed FS 1.6.6 via apt-get method (all modules) and don't see >>> mod_com_g729 anywhere. >>> >>> Should I install it from source ? >>> >>> Please advice. >>> >>> # apt-cache search freeswitch | grep 729 >>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160229/c6e382c1/attachment.html From shlomis at liveperson.com Mon Feb 29 16:43:16 2016 From: shlomis at liveperson.com (Shlomi Schwartz) Date: Mon, 29 Feb 2016 15:43:16 +0200 Subject: [Freeswitch-users] MOD_VERTO Documentation In-Reply-To: References: Message-ID: super, thanks On Sun, Feb 28, 2016 at 5:17 PM, Shlomi Schwartz wrote: > Hi all, > > Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any > documentation regarding the dialogParams on verto.invite method? > > The only documentation I found was : > https://freeswitch.org/confluence/display/FREESWITCH/mod_verto > > Thanks :) > > -- > Shlomi Schwartz > R&D Technical Leader > T: +972-74-700-4511 > > We Create Meaningful Connections > > > -- Shlomi Schwartz R&D Technical Leader T: +972-74-700-4511 We Create Meaningful Connections -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/36cbd39a/attachment.html From ahabiba at gmail.com Mon Feb 29 17:13:06 2016 From: ahabiba at gmail.com (Ahmed Habiba) Date: Mon, 29 Feb 2016 17:13:06 +0300 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: References: Message-ID: Hi, Where is the firewall, is it between user and freeswitch or freeswitch and opensips All cases freeswitch should send the firewall IP for the endpoint behind firewall so that the endpoint would be able to communicate back to freeswitch correct external IP. Thanks, Ahmed Habiba. From: Rutu Patel > Subject: Re: [Freeswitch-users] Call dropping after 32 seconds Date: February 29, 2016 at 4:15:28 PM GMT+3 To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > Hi Jurijs, We have to consider the setup with freeswitch and opensips only. registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) And there is a firewall and we have added x.x.x.3 IP there. Is it possible there is some network related issue Or any sip profile parameter we need to set? -- Thanks, Rutu Patel On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga > wrote: Hi, 1) You have very complex set-up and I doubt that you need it. 2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP. 3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty. With kind regards, Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel >: Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look? Thanks. -- Thanks, Rutu Patel On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus > wrote: 5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel > wrote: Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel >: Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look? Thanks. -- Thanks, Rutu Patel On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus > wrote: 5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel > wrote: Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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/20160229/950adf6f/attachment-0001.html From brian at freeswitch.org Mon Feb 29 17:30:31 2016 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Feb 2016 08:30:31 -0600 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: Are you using a nonstandard install path? On Monday, February 29, 2016, ???? ?????? wrote: > Thanks guys. > I just had to set manually the path for script fs-latest-installer-v1.6 > > Otherwise mod_com_g729.so was not installed anywhere (without any warning > message). > > 2016-02-28 0:00 GMT+03:00 Ken Rice >: > >> I'm pretty sure the link he gave you is the g729 commercial mod >> installation instructions and answers your question >> >> Sent from my iPhone >> >> On Feb 27, 2016, at 12:57 PM, ???? ?????? > > wrote: >> >> Looks like you didn't read my question. >> >> 2016-02-27 18:49 GMT+03:00 ?talo Rossi > >: >> >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 >>> >>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? >> > wrote: >>> >>>> Hi list >>>> >>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't >>>> see mod_com_g729 anywhere. >>>> >>>> Should I install it from source ? >>>> >>>> Please advice. >>>> >>>> # apt-cache search freeswitch | grep 729 >>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 >> > > -- *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/20160229/7876654e/attachment.html From govoiper at gmail.com Mon Feb 29 17:41:23 2016 From: govoiper at gmail.com (SamyGo) Date: Mon, 29 Feb 2016 09:41:23 -0500 Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers Message-ID: Hi All, I'm looking for some guidance on connecting my ESL client program to multiple FS Servers at once. I've written a small code which connects to one FS ESL and based on events it generates some reports and also send back some actions to FS. Now I'm looking to expand that form one to multiple FS servers but the code should collect all events from every FS and be able to send action to any of them whatever the logic be. Any guides or hints will be much appreciated. Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/cdad7425/attachment.html From deforceczt at gmail.com Mon Feb 29 18:23:48 2016 From: deforceczt at gmail.com (Vladislav Ivanov) Date: Mon, 29 Feb 2016 17:23:48 +0200 Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7 Message-ID: Hey guys, I'm unable to compile mod_sndfile even though I have installed all the devel packages. # yum install libsndfile-devel Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and latest version # make mod_sndfile make[1]: Entering directory `/opt/freeswitch' make libfreeswitch.la make[2]: Entering directory `/opt/freeswitch' make[2]: Leaving directory `/opt/freeswitch' make[1]: Leaving directory `/opt/freeswitch' make[1]: Entering directory `/opt/freeswitch/src/mod' make[2]: Entering directory `/opt/freeswitch/src/mod' making all mod_sndfile make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile' Makefile:888: *** You must install libsndfile-dev to build mod_sndfile. Stop. make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile' make[2]: *** [mod_sndfile-all] Error 1 make[2]: Leaving directory `/opt/freeswitch/src/mod' make[1]: *** [mod_sndfile] Error 2 make[1]: Leaving directory `/opt/freeswitch/src/mod' make: *** [mod_sndfile] Error 2 Any tips? Best Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/1835adb8/attachment.html From bobjectsfreeswitch at gmail.com Mon Feb 29 18:29:42 2016 From: bobjectsfreeswitch at gmail.com (Bob Hartwig) Date: Mon, 29 Feb 2016 09:29:42 -0600 Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers In-Reply-To: References: Message-ID: Is that inherently more difficult than connecting to one server? Consider taking an object-oriented approach: Make a FreeswitchServer object with address ivar, have the constructor connect a socket, instantiate once for each server. On Mon, Feb 29, 2016 at 8:41 AM, SamyGo wrote: > Hi All, > > I'm looking for some guidance on connecting my ESL client program to > multiple FS Servers at once. I've written a small code which connects to > one FS ESL and based on events it generates some reports and also send back > some actions to FS. > > Now I'm looking to expand that form one to multiple FS servers but the > code should collect all events from every FS and be able to send action to > any of them whatever the logic be. > > Any guides or hints will be much appreciated. > > Regards, > Sammy > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160229/fd374fd3/attachment-0001.html From abaci64 at gmail.com Mon Feb 29 18:53:54 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 29 Feb 2016 10:53:54 -0500 Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function In-Reply-To: References: Message-ID: That's exactly what I noticed, it breaks out of the current function, why not open a Jira? I don't think it's supposed to behave this way. On Mon, Feb 29, 2016 at 1:37 AM, Andrew Keil wrote: > Thanks for your response. > > > > I have gone through these with no luck. Like I said the > session:destroy(???) crashes FreeSWITCH, which is therefore off the list. > The rest simply interrupt the current function and do no end the script. > > > > I guess my next move is to see why session:destroy() crashes FreeSWITCH, > however I am a little snowed under at the moment so if anyone has some time > to replicate this (only needs one line of code in a Lua script) and pass > this on to the developers that would be great. > > > > Andrew > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B > *Sent:* Saturday, 27 February 2016 1:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Re- End Lua script after HangupHook > handled without all the extra code to handle the return to the function > > > > See > https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook > for a few ways to exit the lua script (error(), return "exit", return > "die", s:destroy("error message")). I personally tried return "exit" but it > seems to me that it only exits the calling function, haven't had a chance > to look further, it's possible that the calling it from the within a > function is different. if you play around and figure out please report back. > > > > On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil > wrote: > > To FreeSWITCH Users, > > > > See below for a sample template for a Lua Service Script running inside > FreeSWITCH. > > > > The issue I have is fairly straightforward. > > > > I need a function to run when hangup is detected (ie. at the end of the > call) however I understand this must not delay ending the script. This > function is CleanUp(). Then I would like the service to end. > > > > The problem I am having is if the caller hangs up during the playback of > ?intro.wav? (as shown inside the MainService() function below), then the > code jumps to the myHangupHook which calls CleanUp() perfectly, the issue > is once CleanUp() is complete I would like the Lua script to end there and > then (ie. at the bottom of CleanUp()). What actually happens is it returns > to MainService() and continues to try and play ?info.wav?, unless I either > check for session:ready() everywhere or add a goto as shown below under > each streamFile() function call. > > > > My aim is to reduce extra code and to make the Lua script simpler and > easier to read. Also I would like to try and avoid goto statements, which > I know can be done with if (session:ready()) etc?. > > > > So is there a way to stop a Lua script running inside FreeSWITCH cleanly? > I have tried the os.exit() this is barred from use by FreeSWITCH. I have > also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on > CentOS 6.7, CentOS 7 and windows) 100% of the time! > > > > I could look further into the Lua additions done by the FreeSWITCH team in > the source code, however if someone has already solved this then that would > be the best solution. > > > > FYI: Obviously the script below is simple, however I am sure that you > understand if the script was complicated having to use ?*if > (session:ready()) then ?.?* or ?*if (not session:ready()) then goto > HANGUPEXIT end?* makes the code ugly. > > > > Thanks in advance, > > > > Andrew Keil > > *Visytel Pty Ltd* > > > > > > > > ------------------------------------------------------------------------------ > Sample Lua Service > ----------------------------------------------------------------------- > > > > -- Lua template for FreeSWITCH service > > -- By: Andrew Keil (Visytel Pty Ltd) > > -- Email: support at visytel.com > > > > -- Setup script wide variables here > > > > function PreAnswer() > > freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); > > -- Add your pre answer code from here > > > > -- End of your pre answer code > > freeswitch.consoleLog("INFO", "PRE ANSWER SECTION > COMPLETE\n"); > > end > > > > function AnswerCaller() > > session:answer() > > session:sleep(1000) > > end > > > > function MainService() > > freeswitch.consoleLog("INFO", "MAIN SERVICE > SECTION\n"); > > if (session:ready()) then > > -- Note (1): If you wish to end the call > then simply use: goto ENDSERVICE > > -- Note (2): To terminate the service > sooner when HANGUP is detected use: if (not session:ready()) then goto > HANGUPEXIT end > > -- Add your main service code from > here (caller would have been answered) > > > > session:streamFile("intro.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > session:streamFile("info.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > > session:streamFile("outro.wav") > > if (not session:ready()) then goto > HANGUPEXIT end > > > > -- End of your main service code > > end > > ::ENDSERVICE:: > > if (session:ready()) then > > -- End of service so hangup > > session:hangup() -- Should automatically > jump to CleanUp() via hangup handler if caller still online at this stage > > end > > goto END > > ::HANGUPEXIT:: > > freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP > DETECTED)\n"); > > ::END:: > > freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION > COMPLETE\n"); > > end > > > > function CleanUp() > > freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); > > -- Add your cleanup code from here (caller would have been > disconnected) > > > > -- End of your cleanup code > > freeswitch.consoleLog("INFO", "CLEANUP SECTION > COMPLETE\n"); > > end > > > > function myHangupHook(s, status, arg) > > session:hangup() > > CleanUp() -- Run CleanUp function now since the caller has > disconnected > > end > > > > -- Setup Hangup event handler here > > v_hangup = "HANGUP" > > session:setHangupHook("myHangupHook", "v_hangup") > > > > -- Call service functions in order > > PreAnswer() > > AnswerCaller() > > MainService() > > -- End of Lua service > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160229/b253cf11/attachment-0001.html From govoiper at gmail.com Mon Feb 29 18:55:13 2016 From: govoiper at gmail.com (SamyGo) Date: Mon, 29 Feb 2016 10:55:13 -0500 Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers In-Reply-To: References: Message-ID: Hi Bob, Thanks for taking out time, I understand what you're saying, and it makes sense. I have used perl so far and I am considering GoLang for this. (more of a scripting approach so far) Can you just tell me this: if it is an ObjOriented code and I connect to multiple servers at once, the buffer I receive Events into will be common ? The same loop in which I collect events from all the servers stays the same too !? Considering that its been some time I've done some good programming, my dev skills are decently rusty and hence this email. Thanks again, Sammy On Mon, Feb 29, 2016 at 10:29 AM, Bob Hartwig wrote: > Is that inherently more difficult than connecting to one server? Consider > taking an object-oriented approach: Make a FreeswitchServer object with > address ivar, have the constructor connect a socket, instantiate once for > each server. > > > > On Mon, Feb 29, 2016 at 8:41 AM, SamyGo wrote: > >> Hi All, >> >> I'm looking for some guidance on connecting my ESL client program to >> multiple FS Servers at once. I've written a small code which connects to >> one FS ESL and based on events it generates some reports and also send back >> some actions to FS. >> >> Now I'm looking to expand that form one to multiple FS servers but the >> code should collect all events from every FS and be able to send action to >> any of them whatever the logic be. >> >> Any guides or hints will be much appreciated. >> >> Regards, >> Sammy >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160229/c6266703/attachment.html From abaci64 at gmail.com Mon Feb 29 19:15:51 2016 From: abaci64 at gmail.com (Abaci B) Date: Mon, 29 Feb 2016 11:15:51 -0500 Subject: [Freeswitch-users] Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function In-Reply-To: References: Message-ID: I don't see in your example above where you have a return "exit" in your hangup hook function see https://freeswitch.org/jira/browse/FS-3841 seems like the "exit" or "die" needs to be returned directly from the hanguphook function so try from that function (not cleanup function) to return "exit" or "dye", also it seems On Mon, Feb 29, 2016 at 10:53 AM, Abaci B wrote: > That's exactly what I noticed, it breaks out of the current function, why > not open a Jira? I don't think it's supposed to behave this way. > > On Mon, Feb 29, 2016 at 1:37 AM, Andrew Keil > wrote: > >> Thanks for your response. >> >> >> >> I have gone through these with no luck. Like I said the >> session:destroy(???) crashes FreeSWITCH, which is therefore off the list. >> The rest simply interrupt the current function and do no end the script. >> >> >> >> I guess my next move is to see why session:destroy() crashes FreeSWITCH, >> however I am a little snowed under at the moment so if anyone has some time >> to replicate this (only needs one line of code in a Lua script) and pass >> this on to the developers that would be great. >> >> >> >> Andrew >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Abaci B >> *Sent:* Saturday, 27 February 2016 1:29 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Re- End Lua script after HangupHook >> handled without all the extra code to handle the return to the function >> >> >> >> See >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:setHangupHook >> for a few ways to exit the lua script (error(), return "exit", return >> "die", s:destroy("error message")). I personally tried return "exit" but it >> seems to me that it only exits the calling function, haven't had a chance >> to look further, it's possible that the calling it from the within a >> function is different. if you play around and figure out please report back. >> >> >> >> On Thu, Feb 25, 2016 at 9:33 PM, Andrew Keil >> wrote: >> >> To FreeSWITCH Users, >> >> >> >> See below for a sample template for a Lua Service Script running inside >> FreeSWITCH. >> >> >> >> The issue I have is fairly straightforward. >> >> >> >> I need a function to run when hangup is detected (ie. at the end of the >> call) however I understand this must not delay ending the script. This >> function is CleanUp(). Then I would like the service to end. >> >> >> >> The problem I am having is if the caller hangs up during the playback of >> ?intro.wav? (as shown inside the MainService() function below), then the >> code jumps to the myHangupHook which calls CleanUp() perfectly, the issue >> is once CleanUp() is complete I would like the Lua script to end there and >> then (ie. at the bottom of CleanUp()). What actually happens is it returns >> to MainService() and continues to try and play ?info.wav?, unless I either >> check for session:ready() everywhere or add a goto as shown below under >> each streamFile() function call. >> >> >> >> My aim is to reduce extra code and to make the Lua script simpler and >> easier to read. Also I would like to try and avoid goto statements, which >> I know can be done with if (session:ready()) etc?. >> >> >> >> So is there a way to stop a Lua script running inside FreeSWITCH >> cleanly? I have tried the os.exit() this is barred from use by >> FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH >> (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! >> >> >> >> I could look further into the Lua additions done by the FreeSWITCH team >> in the source code, however if someone has already solved this then that >> would be the best solution. >> >> >> >> FYI: Obviously the script below is simple, however I am sure that you >> understand if the script was complicated having to use ?*if >> (session:ready()) then ?.?* or ?*if (not session:ready()) then goto >> HANGUPEXIT end?* makes the code ugly. >> >> >> >> Thanks in advance, >> >> >> >> Andrew Keil >> >> *Visytel Pty Ltd* >> >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> Sample Lua Service >> ----------------------------------------------------------------------- >> >> >> >> -- Lua template for FreeSWITCH service >> >> -- By: Andrew Keil (Visytel Pty Ltd) >> >> -- Email: support at visytel.com >> >> >> >> -- Setup script wide variables here >> >> >> >> function PreAnswer() >> >> freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); >> >> -- Add your pre answer code from here >> >> >> >> -- End of your pre answer code >> >> freeswitch.consoleLog("INFO", "PRE ANSWER SECTION >> COMPLETE\n"); >> >> end >> >> >> >> function AnswerCaller() >> >> session:answer() >> >> session:sleep(1000) >> >> end >> >> >> >> function MainService() >> >> freeswitch.consoleLog("INFO", "MAIN SERVICE >> SECTION\n"); >> >> if (session:ready()) then >> >> -- Note (1): If you wish to end the call >> then simply use: goto ENDSERVICE >> >> -- Note (2): To terminate the service >> sooner when HANGUP is detected use: if (not session:ready()) then goto >> HANGUPEXIT end >> >> -- Add your main service code from >> here (caller would have been answered) >> >> >> >> session:streamFile("intro.wav") >> >> if (not session:ready()) then goto >> HANGUPEXIT end >> >> session:streamFile("info.wav") >> >> if (not session:ready()) then goto >> HANGUPEXIT end >> >> >> session:streamFile("outro.wav") >> >> if (not session:ready()) then goto >> HANGUPEXIT end >> >> >> >> -- End of your main service code >> >> end >> >> ::ENDSERVICE:: >> >> if (session:ready()) then >> >> -- End of service so hangup >> >> session:hangup() -- Should automatically >> jump to CleanUp() via hangup handler if caller still online at this stage >> >> end >> >> goto END >> >> ::HANGUPEXIT:: >> >> freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP >> DETECTED)\n"); >> >> ::END:: >> >> freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION >> COMPLETE\n"); >> >> end >> >> >> >> function CleanUp() >> >> freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); >> >> -- Add your cleanup code from here (caller would have >> been disconnected) >> >> >> >> -- End of your cleanup code >> >> freeswitch.consoleLog("INFO", "CLEANUP SECTION >> COMPLETE\n"); >> >> end >> >> >> >> function myHangupHook(s, status, arg) >> >> session:hangup() >> >> CleanUp() -- Run CleanUp function now since the caller >> has disconnected >> >> end >> >> >> >> -- Setup Hangup event handler here >> >> v_hangup = "HANGUP" >> >> session:setHangupHook("myHangupHook", "v_hangup") >> >> >> >> -- Call service functions in order >> >> PreAnswer() >> >> AnswerCaller() >> >> MainService() >> >> -- End of Lua service >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160229/150eb38a/attachment-0001.html From bobjectsfreeswitch at gmail.com Mon Feb 29 19:34:35 2016 From: bobjectsfreeswitch at gmail.com (Bob Hartwig) Date: Mon, 29 Feb 2016 10:34:35 -0600 Subject: [Freeswitch-users] ESL Client --> Multiple FS Servers In-Reply-To: References: Message-ID: I guess it depends on whether you need to send commands in reply to events. If so, you'll probably want to have separate queues and servicing threads for each server. Your requirements can probably tell you better than I can. Bob On Mon, Feb 29, 2016 at 9:55 AM, SamyGo wrote: > Hi Bob, > Thanks for taking out time, I understand what you're saying, and it makes > sense. I have used perl so far and I am considering GoLang for this. (more > of a scripting approach so far) > > Can you just tell me this: if it is an ObjOriented code and I connect to > multiple servers at once, the buffer I receive Events into will be common ? > The same loop in which I collect events from all the servers stays the same > too !? > > Considering that its been some time I've done some good programming, my > dev skills are decently rusty and hence this email. > > Thanks again, > Sammy > > On Mon, Feb 29, 2016 at 10:29 AM, Bob Hartwig < > bobjectsfreeswitch at gmail.com> wrote: > >> Is that inherently more difficult than connecting to one server? >> Consider taking an object-oriented approach: Make a FreeswitchServer >> object with address ivar, have the constructor connect a socket, >> instantiate once for each server. >> >> >> >> On Mon, Feb 29, 2016 at 8:41 AM, SamyGo wrote: >> >>> Hi All, >>> >>> I'm looking for some guidance on connecting my ESL client program to >>> multiple FS Servers at once. I've written a small code which connects to >>> one FS ESL and based on events it generates some reports and also send back >>> some actions to FS. >>> >>> Now I'm looking to expand that form one to multiple FS servers but the >>> code should collect all events from every FS and be able to send action to >>> any of them whatever the logic be. >>> >>> Any guides or hints will be much appreciated. >>> >>> Regards, >>> Sammy >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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/20160229/b164afaf/attachment.html From Shawn.Wheeler at interlockconcepts.com Mon Feb 29 08:18:17 2016 From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler) Date: Mon, 29 Feb 2016 05:18:17 +0000 Subject: [Freeswitch-users] Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit and service startup Message-ID: Parameters Virtual Machine running Windows Server 2012 R2 base machine. I have installed FreeSWITCH-1.7.0-0a024c4ecb-64bit I have run into two issues as noted below. Any help would be most appreciated. Item 1 When I tried to start the service I received the following error. ?Windows could not start the FreeSWITCH Multi Protocol Switch service on Local Computer Error 1053: The service did not respond to start or control request in a timely fashion? Item 2 I also received this error when trying to start the CLI ?The program can't stat because VCRUNTIME140.dll is missing from your computer. Try reinstalling the program to fix this problem.? I assume it was due to the service not starting. Thank you all in advance. Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/ddbbf6e5/attachment.html From delvin.friends at gmail.com Mon Feb 29 21:46:38 2016 From: delvin.friends at gmail.com (Delvin Varghese) Date: Mon, 29 Feb 2016 18:46:38 +0000 Subject: [Freeswitch-users] Recording each leg of a conference call separately Message-ID: <4567C63B-8AF4-40A4-B9F5-915E4073F1CF@gmail.com> Hi all, I am familiar with the record and record_session commands in freeswitch to initiate recordings, and also using ESL.. But is there a way to record ONLY the speech from one device at a time in separate recordings? The way i am envisioning is that if there are 9 different devices involved in a conference call, that at the end of it I would be able to access 9 recordings (each containing only the outbound voice from that device)? Is this possible? What are people?s thoughts? Delvin From yehavi.bourvine at gmail.com Mon Feb 29 21:55:27 2016 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 29 Feb 2016 20:55:27 +0200 Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7 In-Reply-To: References: Message-ID: If you do not need this module then simply comment it out in modules.conf; that's what I did... __Yehavi: 2016-02-29 17:23 GMT+02:00 Vladislav Ivanov : > Hey guys, > > I'm unable to compile mod_sndfile even though I have installed all the > devel packages. > > # yum install libsndfile-devel > Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and latest > version > > # make mod_sndfile > make[1]: Entering directory `/opt/freeswitch' > make libfreeswitch.la > make[2]: Entering directory `/opt/freeswitch' > make[2]: Leaving directory `/opt/freeswitch' > make[1]: Leaving directory `/opt/freeswitch' > make[1]: Entering directory `/opt/freeswitch/src/mod' > make[2]: Entering directory `/opt/freeswitch/src/mod' > > making all mod_sndfile > make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile' > Makefile:888: *** You must install libsndfile-dev to build mod_sndfile. > Stop. > make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile' > make[2]: *** [mod_sndfile-all] Error 1 > make[2]: Leaving directory `/opt/freeswitch/src/mod' > make[1]: *** [mod_sndfile] Error 2 > make[1]: Leaving directory `/opt/freeswitch/src/mod' > make: *** [mod_sndfile] Error 2 > > > Any tips? > 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/20160229/672d4cef/attachment-0001.html From krice at freeswitch.org Mon Feb 29 21:57:15 2016 From: krice at freeswitch.org (Ken Rice) Date: Mon, 29 Feb 2016 12:57:15 -0600 Subject: [Freeswitch-users] Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit and service startup In-Reply-To: References: Message-ID: <42c401d17323$024793f0$06d6bbd0$@freeswitch.org> Just install the VC runtime from microsoft From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler Sent: Sunday, February 28, 2016 11:18 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit and service startup Parameters Virtual Machine running Windows Server 2012 R2 base machine. I have installed FreeSWITCH-1.7.0-0a024c4ecb-64bit I have run into two issues as noted below. Any help would be most appreciated. Item 1 When I tried to start the service I received the following error. ?Windows could not start the FreeSWITCH Multi Protocol Switch service on Local Computer Error 1053: The service did not respond to start or control request in a timely fashion? Item 2 I also received this error when trying to start the CLI ?The program can't stat because VCRUNTIME140.dll is missing from your computer. Try reinstalling the program to fix this problem.? I assume it was due to the service not starting. Thank you all in advance. Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/f01c43da/attachment.html From ynasida at gmail.com Mon Feb 29 22:01:25 2016 From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=) Date: Mon, 29 Feb 2016 22:01:25 +0300 Subject: [Freeswitch-users] segfaults because odbc mysql driver Message-ID: Hi guys, As far as I know there is one already known problems with FS segfaults that happen because of odbc mysql driver. I have latest recommended setup with debian 8.3 + FS 1.6. Somebody mentioned that version of unixODBC should be higher than 2.3 to solve this but... I have unixODBC 2.3.1 and it doesn't help. Everybody use pgsql ? :) Please advice if somebody use mysql with FS 1.6 and still happy. I also have backtrace file here - https://freeswitch.org/jira/browse/FS-8857 and will be glad any ideas. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160229/964c2b41/attachment.html From deforceczt at gmail.com Mon Feb 29 22:05:45 2016 From: deforceczt at gmail.com (Vladislav Ivanov) Date: Mon, 29 Feb 2016 21:05:45 +0200 Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7 In-Reply-To: References: Message-ID: Yeah, I need it... 2016-02-29 20:55 GMT+02:00 Yehavi Bourvine : > If you do not need this module then simply comment it out in modules.conf; > that's what I did... > > __Yehavi: > > 2016-02-29 17:23 GMT+02:00 Vladislav Ivanov : > >> Hey guys, >> >> I'm unable to compile mod_sndfile even though I have installed all the >> devel packages. >> >> # yum install libsndfile-devel >> Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and >> latest version >> >> # make mod_sndfile >> make[1]: Entering directory `/opt/freeswitch' >> make libfreeswitch.la >> make[2]: Entering directory `/opt/freeswitch' >> make[2]: Leaving directory `/opt/freeswitch' >> make[1]: Leaving directory `/opt/freeswitch' >> make[1]: Entering directory `/opt/freeswitch/src/mod' >> make[2]: Entering directory `/opt/freeswitch/src/mod' >> >> making all mod_sndfile >> make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile' >> Makefile:888: *** You must install libsndfile-dev to build mod_sndfile. >> Stop. >> make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile' >> make[2]: *** [mod_sndfile-all] Error 1 >> make[2]: Leaving directory `/opt/freeswitch/src/mod' >> make[1]: *** [mod_sndfile] Error 2 >> make[1]: Leaving directory `/opt/freeswitch/src/mod' >> make: *** [mod_sndfile] Error 2 >> >> >> Any tips? >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20160229/341fa6ce/attachment.html From govoiper at gmail.com Mon Feb 29 22:09:51 2016 From: govoiper at gmail.com (SamyGo) Date: Mon, 29 Feb 2016 14:09:51 -0500 Subject: [Freeswitch-users] Unable to compile mod_sndfile on CentOS 7 In-Reply-To: References: Message-ID: Hi, I recently got into trouble where I compiled FS on ubuntu 14.04 and got error on this module - so I commented it out. Turned out that I was unable to play .wav files in FS. Hence I had to compile this module using an older version of libsndfile-dev since Ubuntu 14.04 installed libsndfile1.dev. It seems like you already have the same version which worked for me, maybe try executing command "ldconfig" on shell and then try. Thanks Sammy On Mon, Feb 29, 2016 at 2:05 PM, Vladislav Ivanov wrote: > Yeah, I need it... > > 2016-02-29 20:55 GMT+02:00 Yehavi Bourvine : > >> If you do not need this module then simply comment it out in >> modules.conf; that's what I did... >> >> __Yehavi: >> >> 2016-02-29 17:23 GMT+02:00 Vladislav Ivanov : >> >>> Hey guys, >>> >>> I'm unable to compile mod_sndfile even though I have installed all the >>> devel packages. >>> >>> # yum install libsndfile-devel >>> Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and >>> latest version >>> >>> # make mod_sndfile >>> make[1]: Entering directory `/opt/freeswitch' >>> make libfreeswitch.la >>> make[2]: Entering directory `/opt/freeswitch' >>> make[2]: Leaving directory `/opt/freeswitch' >>> make[1]: Leaving directory `/opt/freeswitch' >>> make[1]: Entering directory `/opt/freeswitch/src/mod' >>> make[2]: Entering directory `/opt/freeswitch/src/mod' >>> >>> making all mod_sndfile >>> make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile' >>> Makefile:888: *** You must install libsndfile-dev to build mod_sndfile. >>> Stop. >>> make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile' >>> make[2]: *** [mod_sndfile-all] Error 1 >>> make[2]: Leaving directory `/opt/freeswitch/src/mod' >>> make[1]: *** [mod_sndfile] Error 2 >>> make[1]: Leaving directory `/opt/freeswitch/src/mod' >>> make: *** [mod_sndfile] Error 2 >>> >>> >>> Any tips? >>> 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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20160229/465cf8cd/attachment-0001.html From jurij.ivo at gmail.com Mon Feb 29 22:27:30 2016 From: jurij.ivo at gmail.com (Jurijs Ivolga) Date: Mon, 29 Feb 2016 21:27:30 +0200 Subject: [Freeswitch-users] Call dropping after 32 seconds In-Reply-To: References: <0000015307b150a5-132a41d6-5bea-4af1-9686-2653df68a3b6-000000@email.amazonses.com> Message-ID: Hi Rutu, Firewall should not be an issue. If you think that it may change SIP packets, you can always use TLS. I doubt that it is good idea to add one more server in set-up, just because of firewall, you just need to configure all of your servers, devices properly. If you can, you should eliminate unnecessary servers from your set-up. >From what you described before, it might be issue not connected to NAT, but because Opensips wasn't configured properly. I had similar issue when Kamailio(Opensip is fork from Kamailio project, so they almost identical) was wrongly configured, particularly path header wasn't inserted by Kamailio. But this 30 seconds timeout is quite often NAT issue, but again if you have NAT issue, you should not blame FW or anything else, you should just configure your Freeswitch and Opensips properly. Almost all devices in internet are behind NAT and almost all of them works perfectly with VoIP. So if you need, help, then please send full sip trace, so I can take a look on it. With kind regards, Jurijs 2016-02-29 15:15 GMT+02:00 Rutu Patel : > Hi Jurijs, > > We have to consider the setup with freeswitch and opensips only. > registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips > server) > > And there is a firewall and we have added x.x.x.3 IP there. > > Is it possible there is some network related issue Or any sip profile > parameter we need to set? > > -- > Thanks, > Rutu Patel > > On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga > wrote: > >> Hi, >> >> 1) You have very complex set-up and I doubt that you need it. >> >> 2) As far as you have user with ip x.x.x.174 and opensips server with >> same ip x.x.x.174 it very hard to debug. So I propose you to send new log >> where will be difference between user ip and opensips IP. >> >> 3) If you have possibility, try to register directly with a user to >> x.x.x.3 gateway and check if same issue still exists, if there is no such >> issue anymore, then thee is definitely issue in your opensips x.x.x.174 and >> freeswitch x.x.x.166. My point here is that you need to isolate issue and >> to understand what part of your set-up works as expected and what is faulty. >> >> With kind regards, >> >> Jurijs >> >> 2016-02-23 7:51 GMT+02:00 Rutu Patel : >> >>> Thanks for the reply. >>> >>> Got your point about NATing issue and no response of 200 OK and as a >>> resoult ACK Timeout. >>> So, now to resolve the issue, if you can assist, what could be the >>> possible fixies? >>> From where can i start? where to look? >>> >>> Thanks. >>> >>> -- >>> Thanks, >>> Rutu Patel >>> >>> >>> >>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: >>> >>>> 5 second response: 32 seconds is a timer/[network/NAT] issue. >>>> >>>> You have lots of 200s to the user since it's waiting for an ACK and >>>> keeps retrying, but for whatever network reason (router... sip alg?), it >>>> isn't getting one, so it triggers a timer to stop the call. >>>> >>>> >>>> -Avi Marcus >>>> BestFone >>>> >>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> Having issue of call dropping after 32 seconds, here are the details- >>>>> >>>>> x.x.x.174: opensips server >>>>> x.x.x.166: freeswitch server >>>>> x.x.x.3: another opensips server which is registered as gateway on >>>>> above freeswitch server >>>>> x.x.x.6: freeswitch server >>>>> x.x.x.47: server through which the user is registered >>>>> I am trying to call from xxxx9 to xxxxxxx29858 >>>>> xxxxxxx00181 is caller-id name and caller-id number >>>>> >>>>> Call flow is like this: >>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 >>>>> (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>>>> >>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>>>> >>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>>>> 2) Call hit the freeswitch server x.x.x.166 >>>>> 3) After '180 Ringing' and '183 Session Progress' packet >>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>>>> gateway x.x.x.3 >>>>> But after 32 seconds call is dropped, >>>>> Within 32 seconds audio is ok from both end so it should not be the >>>>> RTP issue. >>>>> Here I have attached the file with sip logs, you can observer from the >>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>>>> >>>>> What is wrong here? Any help would be appreciated here. >>>>> >>>>> Here is the file with sip logs >>>>> -- >>>>> Thanks, >>>>> Rutu Patel >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>> >> >> >> Jurijs >> >> 2016-02-23 7:51 GMT+02:00 Rutu Patel : >> >>> Thanks for the reply. >>> >>> Got your point about NATing issue and no response of 200 OK and as a >>> resoult ACK Timeout. >>> So, now to resolve the issue, if you can assist, what could be the >>> possible fixies? >>> From where can i start? where to look? >>> >>> Thanks. >>> >>> -- >>> Thanks, >>> Rutu Patel >>> >>> >>> >>> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus wrote: >>> >>>> 5 second response: 32 seconds is a timer/[network/NAT] issue. >>>> >>>> You have lots of 200s to the user since it's waiting for an ACK and >>>> keeps retrying, but for whatever network reason (router... sip alg?), it >>>> isn't getting one, so it triggers a timer to stop the call. >>>> >>>> >>>> -Avi Marcus >>>> BestFone >>>> >>>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> Having issue of call dropping after 32 seconds, here are the details- >>>>> >>>>> x.x.x.174: opensips server >>>>> x.x.x.166: freeswitch server >>>>> x.x.x.3: another opensips server which is registered as gateway on >>>>> above freeswitch server >>>>> x.x.x.6: freeswitch server >>>>> x.x.x.47: server through which the user is registered >>>>> I am trying to call from xxxx9 to xxxxxxx29858 >>>>> xxxxxxx00181 is caller-id name and caller-id number >>>>> >>>>> Call flow is like this: >>>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 >>>>> (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) >>>>> >>>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration. >>>>> >>>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 >>>>> 2) Call hit the freeswitch server x.x.x.166 >>>>> 3) After '180 Ringing' and '183 Session Progress' packet >>>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the >>>>> gateway x.x.x.3 >>>>> But after 32 seconds call is dropped, >>>>> Within 32 seconds audio is ok from both end so it should not be the >>>>> RTP issue. >>>>> Here I have attached the file with sip logs, you can observer from the >>>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 >>>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then >>>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. >>>>> >>>>> What is wrong here? Any help would be appreciated here. >>>>> >>>>> Here is the file with sip logs >>>>> -- >>>>> Thanks, >>>>> Rutu Patel >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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/20160229/d4acaf2f/attachment-0001.html From ynasida at gmail.com Mon Feb 29 23:00:27 2016 From: ynasida at gmail.com (=?UTF-8?B?0K7RgNC40Lkg0J3QsNGB0LjQtNCw?=) Date: Mon, 29 Feb 2016 23:00:27 +0300 Subject: [Freeswitch-users] where mod_com_g729 ? In-Reply-To: References: Message-ID: no, I installed all things using apt-get and did not set nonstandard path anywhere. 29 ??? 2016 ?. 17:31 ???????????? "Brian West" ???????: > Are you using a nonstandard install path? > > On Monday, February 29, 2016, ???? ?????? wrote: > >> Thanks guys. >> I just had to set manually the path for script fs-latest-installer-v1.6 >> >> Otherwise mod_com_g729.so was not installed anywhere (without any warning >> message). >> >> 2016-02-28 0:00 GMT+03:00 Ken Rice : >> >>> I'm pretty sure the link he gave you is the g729 commercial mod >>> installation instructions and answers your question >>> >>> Sent from my iPhone >>> >>> On Feb 27, 2016, at 12:57 PM, ???? ?????? wrote: >>> >>> Looks like you didn't read my question. >>> >>> 2016-02-27 18:49 GMT+03:00 ?talo Rossi : >>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 >>>> >>>> On Sat, Feb 27, 2016 at 5:36 AM, ???? ?????? wrote: >>>> >>>>> Hi list >>>>> >>>>> I have installed FS 1.6.6 via apt-get method (all modules) and don't >>>>> see mod_com_g729 anywhere. >>>>> >>>>> Should I install it from source ? >>>>> >>>>> Please advice. >>>>> >>>>> # apt-cache search freeswitch | grep 729 >>>>> freeswitch-mod-g729 - mod_g729 for FreeSWITCH >>>>> freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug) >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 >>> >> >> > > -- > > *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/20160229/4e489008/attachment.html