February 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

loopback_initial_codec
by Serge S. Yuriev 21 Feb '18

21 Feb '18

Hello, Using FS release 1.6.20 I'm trying to originate call via loopback with preset codec originate {loopback_initial_codec=PCMA}loopback/12550/internal &park() But codec ignored and set to L16. 2018-02-16 18:51:48.294389 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel loopback/12550/internal-a [9de6c978-4ef8-4f2d-a7d8-f630bebe13af] 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 loopback/12550/internal-a setup codec L16/8000/20 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1102 Rename Channel loopback/12550/internal-a->loopback/12550-a [9de6c978-4ef8-4f2d-a7d8-f630bebe13af] 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:1174 (loopback/12550-a) State Change CS_NEW -> CS_INIT 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:601 loopback/12550-a CHANNEL KILL 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:584 (loopback/12550-a) Running State Change CS_INIT (Cur 1 Tot 1230) 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:627 (loopback/12550-a) State INIT 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel loopback/12550-b [e945099e-2ed9-4c49-9d48-58a25756fb08] 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 loopback/12550-b setup codec L16/8000/20 -- Serge S. Yuriev Lead VoIP engineer

2 2

CALL DOES NOT EXIST error in mod_verto
by Muhammad Umair 20 Feb '18

20 Feb '18

Hi, I have built Freeswitch on window 8 (64bit) using Visual Studio. Every time I try to join a conference it gives me following error in Javascript: INVALID METHOD OR NON-EXISTANT CALL REFERENCE IGNORED verto.clientReady I have attached FS logs as well. Any ideas? - Umair

1 0

NAT port problem
by Sean Devoy 20 Feb '18

20 Feb '18

HI all, I have a NAT problem I can't seem to get past. The phone is NATed. From my location with a FIOS router it works fine. But the client recently upgraded to a new SonicWall router and splat, this device failed. There are about 10 Cisco phones at this location working fine. This phone is a Panasonic KX-TGP550. I have poured over the sip logs and identified the problem. The sip packets come from the router IP and port 15xxx but are returned to that IP and the phone's source port 5060 instead of 15xxx. I have tried every NDLB feature I can find, but not every combination. The responses always go back to the wrong port. Any help would be greatly appreciated. Thanks, Sean Sample sip dump: ------------------------------------------------------------------------ recv 492 bytes from udp/[98.204.241.22]:15183 at 23:16:33.764217: ------------------------------------------------------------------------ REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b Max-Forwards: 70 To: <sip:34@XYZ.bizfocused.com> From: <sip:34@XYZ.bizfocused.com>;tag=3263080642 Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179(a)192.168.2.109<mailto:97742e94-d1dc14228bad9baadd810080f041d179@192.168.2.109> CSeq: 1 REGISTER Contact: <sip:34@192.168.2.109:5074> Expires: 3600 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179) Content-Length: 0 ------------------------------------------------------------------------ send 669 bytes to udp/[98.204.241.22]:5074 at 23:16:33.764394: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized

4 7

Syslog forwarding instead of file
by Kevin Olbrich 20 Feb '18

20 Feb '18

Hi! We are currently testing Graylog2 to merge logs from several UC nodes. Is Freeswitch able to send syslog messages to a remote server? Kind regards, Kevin

2 1

The time on FreeSWITCH is faster than server time
by 王聡 20 Feb '18

20 Feb '18

Hey all, Recently I noticed a clock problem on my FS server, the time in fs_cli had about 10sec faster than server time(correct time), so I executed sync_clock to correct it. This server had run about 1 month, and I wonder if I need to make sync_clock per month, and how to fix this clock problem. Regards.

4 3

Stop Muted tone_stream When Unmuted
by Bill McAndrews 20 Feb '18

20 Feb '18

Hello, I am having trouble with users who mute themselves in conference but don't realize they have. These incidents can be from a minute or two to several hours. We want to play an intermittent beep tone in the background to remind them they are muted. I set muted-sound to to a continuous tone_stream loop which serves the purpose. The problem is that when unmuted, the tone continues. I have unmuted-sound set and I've tried uuid_break as well as other sound related commands but the tone continues. Is there a way to stop a tone_stream that I'm missing? If not, is there a way to code the tone stream in muted-sound that will stop when the channel is unmuted? FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit) on Debian 8 Thank-you in advance. Bill -- Bill McAndrews *Infrastructure Administrator* *Direct: 331-684-8823* RemoteLink.com <http://www.remotelink.com/> RemoteLink.com.ph <http://www.remotelink.com.ph/> <http://www.remotelink.com/> <http://www.facebook.com/remotelink> <http://www.twitter.com/remotelinkinc> <http://www.linkedin.com/company/147838?trk=tyah> <http://www.rltalkscloud.wordpress.com/feed> *Specialties:* *Cloud Computing, SugarCRM, Google Apps for Business, SugarCRM 6.0 Quickbooks Integration, Telecommunications, SIP Networks, Java, Php Development, Search Marketing SEO/SEM, Social Media Coaching*

1 0

How to unregister verto user
by David Wafula 20 Feb '18

20 Feb '18

Hi all, I have a verto user who keeps closing browser window without logging out and i need to force unregistration periodically. What command can be used to accomplish this? The only commands am able to use are: verto status/xmlstatus Kind regards --

7 13

"fsctl recover" doesn't work on Freeswitch V1.6.19
by Kevin Chi 19 Feb '18

19 Feb '18

hello everyone, The steps of my call recovery test as below. 1.First I added the line "<param name="track-calls" value="true"/>" to internal.xml. 2.UA called 9664 to play the hold music, and then run "fsctl crash" command. 3.start freeswtich and run "fsctl recover" command. 4.call was interrupted, recover failed. I opened siptrace before "fsctl recover", the sipflow make me can not understand. freeswitch send INVITE to UA freeswitch recive 200 from UA freeswitch send BYE to UA freeswitch recive 200 from UA freeswitch send ACK from UA My freeswitch is only single server, the version is V1.6.19. I have tried the newer version V1.6.20, but the result was same. What's the reason? Pls give me some suggestion, thx a lot. ------------------ With regards, Kevin Chi

4 3

Javascript - session_in_hangup_hook: How?
by Mark White 19 Feb '18

19 Feb '18

Hi - I'm starting to use Freeswitch to receive inbound faxes - which works brilliantly - but I am trying to get my head around how to launch some .js code after hangup, i.e. to deal with the fax file itself. Specifically, I'm expecting that I'm able to access session data in the script by setting session_in_hangup_hook=true, as below. Now, invoking .js files seems to work fine - either in the dialpan itself, or via the hangup hook. But what I can't seem to understand is how the session data is accessible in the invoked javascript. I'm getting the following error on the FS console from hangup.js: Exception: ReferenceError: request is not defined (near: "var env = request.dumpENV("text");") XML Dialplan and hangup.js code fragments below. Any clues as to what I'm doing wrong? (Environment is a current 1.6, Debian 8, installed via FS packages - not from source. mod_v8 is loaded - the scripts are executing, apparently - but I'm stuck trying to find the example code to access the session variables). Any help appreciated, Mark ----- Dialplan XML: <extension name="fax_receive"> <condition field="destination_number" expression="^5551212$"> <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=jsrun /home/mark/hangup.js"/> <action application="answer" /> <action application="javascript" data="/home/mark/postfax.js"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> // hangup.js code fragment console_log("notice","Inside the hangup.js script"); // This executes, I get the console message post-hangup // Get and log all available session variables. var env; env = request.dumpENV("text"); console_log(env);

1 0

Media Timeout disconnect with Verto
by Oleg Stolyar 16 Feb '18

16 Feb '18

Hi guys, what is the verto equivalent of rtp-timeout-sec param? How do I control when to disconnect a verto call when there is no media coming from the browser client?

3 3