- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Detecting the origin of voice activity using VAD
by Cameron Sorlie 05 Mar '09

05 Mar '09

In a sense, you might say I did futz with mod_vmd ... to create mod_vad. There appeared to be just no (easy) way to modify the internal VAD code in the FreeSWITCH core (see switch_rtp.c) to identify the origins of voice activity. And rather than build into mod_vmd, which is a special purpose tool, a separate module for VAD seemed like a reasonable idea. In short, the mod_vad which I've written up independently monitors the read and the write legs of the session it is run on, and tags each VAD_TALK and VAD_NOTALK event it fires with a user-supplied identification marker (a short string) for the leg which the event relates to. At the moment, the VAD algorithm is dead simple, and is much like the one in the core. I will be happy to submit this module, in a little while, after I've had a chance to make it perhaps a bit more useable outside of our own application context. Cam On Mon, Mar 2, 2009 at 5:43 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > > On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale > <anthony.minessale(a)gmail.com> wrote: > > i think that's what mod_vmd does > > > I think that's right. It just does the opposite - instead of looking > for differing power levels it looks for the same power level. In other > words it tries to detect distinctly non-human sound. I'll bet you > could futz with that code and have it fire off events when it detects > what it believes is human speech. > > -MC >

2 1

Patch for mod_event_socket to fire real CUSTOM events via sendevent command
by Helmut Kuper 05 Mar '09

05 Mar '09

Hello, I enhanced mod_event_socket's sendevent command to fire real CUSTOM events with correct subclass and trailing custom headers, so that subscribed nodes can receive those events. Is there a way to get the patch into trunk? regards Helmut

2 1

SIP registration/retry/authorization problem
by Jesse Peterson 05 Mar '09

05 Mar '09

Hello, I seem to be experiencing the exact same issue as is documented here: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/00627… Like above a "sofia profile external restart" immediately resumes operation. Does anyone have an idea what this may be? Is there a debugging step I can take? If it were a predictable outage I could monitor the registration attempts and find out why there are [401][Unauthorized] errors suddenly (again after a 'sofia restart' all is well). This happens multiple times a day for us. Thanks, - Jesse

5 8

Re: [Freeswitch-users] separate gateways SIP and RTP stream
by Justin Miller 04 Mar '09

04 Mar '09

Ok, got it. Thank you. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists@avgs.ca] Sent: Wednesday, March 4, 2009 12:26 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. On 4-Mar-09, at 2:25 PM, Justin Miller wrote: So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists@avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1 0

How to send CUSTOM event via event socket?
by Helmut Kuper 04 Mar '09

04 Mar '09

Hello, how to send a CUSTOM event via event socket? Currently I send this: sendevent CUSTOM\n Event-Name: CUSTOM\n Event-Subclass: myevent::snom-aurl\n Snom-Event: OFFHOOK\n \n I got an +OK reply from FS. I subscribed to ALL event and get only this: Content-Length: 516 Content-Type: text/event-plain Event-Name: COMMAND Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d FreeSWITCH-Hostname: ippbx-prod-node0 FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-03-04%2017%3A42%3A35 Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT Event-Date-Timestamp: 1236184955199959 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1045 Command: sendevent%20CUSTOM Event-Name: CUSTOM Event-Subclass: ewetel%3A%3Asnom-aurl Snom-Event: OFFHOOK But there is no CUSTOM event received. regards helmut

2 1

Re: [Freeswitch-users] separate gateways SIP and RTP stream
by Justin Miller 04 Mar '09

04 Mar '09

So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists@avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

2 1

separate gateways SIP and RTP stream
by Justin Miller 04 Mar '09

04 Mar '09

I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin

2 1

howto master logging in fs_cli
by Cavalera Claudio Luigi 04 Mar '09

04 Mar '09

Hello, I'm trying to clarify behaviour of fs_cli http://wiki.freeswitch.org/wiki/Fs_cli After some experiments I'm still not sure on how to deal with logging. For example: --- root@lallobox:/usr/local/freeswitch/bin# ./fs_cli -d 6 _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ***************************************************** * Anthony Minessale II, Ken Rice, Michael Jerris * * FreeSWITCH (http://www.freeswitch.org) * * Brought to you by ClueCon http://www.cluecon.com/ * ***************************************************** Type /help <enter> to see a list of commands [INFO] libs/esl/fs_cli.c:726 main() FS CLI Ready. enter /help for a list of commands. freeswitch@internal> --- So I assume now I'm logging at level 6 in fs_cli, instead I still see debug messages like this as in loglevel 7: 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1856 listener_run() Connection Open from 127.0.0.1:48400 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1979 listener_run() Session complete, waiting for children To achieve a info loglevel i have to start fs_cli like this ./fs_cli -l info or type /log info in the fs_cli console So could you please someone more expert with me clarify the difference between -d and -l options so that I can update the wiki which is now wrong/incomplete ? -l, --loglevel=command Log Level -q, --quiet Disable logging -d, --debug=level Debug Level (0 - 7) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

3 5

Ghost Sessions in CLI after a longterm test
by rod 04 Mar '09

04 Mar '09

Hi All, I ran some longer tests with FS 1.0.3 acting as an SBC. The test machine has the following specs: - Intel Quad Core Q9550 - 8GB RAM (far too much from what I saw) After 3 days running SIPP with 750 simultaneous calls (1500 channels) at 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. In the CLI, using status command I got this: freeswitch@internal> status UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, 607 microseconds 15817560 session(s) since startup 22 session(s) 0/500 But when I use "show channels" or "show calls", I see nothing. So I'm wondering where are these 22 sessions ? FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. Successful call --> 5271434 Failed call ---> 1554 (less than 0.03%) regards, rod. complete SIPP summary: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 10.10.10.254:5060(UDP) 0 new calls during 0.856 s period 7 ms scheduler resolution 0 calls (limit 750) Peak was 750 calls, after 15 s 0 Running, 34 Paused, 0 Woken up 15544 out-of-call msg (discarded) 1 open sockets 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg INVITE ----------> 5273022 0 0 100 <---------- 5273022 0 1554 180 <---------- 0 0 0 183 <---------- 0 0 0 200 <---------- E-RTD1 5271434 0 0 ACK ----------> 5271434 0 Pause [ 35.0s] 5271434 0 BYE ----------> 5271434 0 0 200 <---------- 5271434 0 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2009-02-27 09:11:31 Last Reset Time | 2009-03-02 07:49:10 Current Time | 2009-03-02 07:49:11 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:857 | 70:37:39:429 Call Rate | 0.000 cps | 20.739 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 5273022 Total Call created | | 5273022 Current Call | 34 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 5271434 Failed call | 0 | 1554 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000 | 00:00:00:240 Call Length | 38:32:13:386 | 00:00:36:131 ------------------------------ Test Terminated --------------------------------

4 5

Mandatory IE missing
by Ali Al-Rubaie 04 Mar '09

04 Mar '09

Hi, During proxy authentication, I got "mandatory IE missing" error in the response as shown below. How this error can be resolved? recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: ------------------------------------------------------------------------ INVITE sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Contact: sip:1002@209.82.10.250:1091 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE Content-Length: 187 Content-Type: application/sdp User-Agent: HelpCaster SoftPHONE Supported: com.hearme.mux v=0 o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 s=HelpCaster c=IN IP4 209.82.10.250 t=0 0 m=audio 8002 RTP/AVP 0 4 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Content-Length: 0 ------------------------------------------------------------------------ send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=K83mZmZH8g3Hr Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="209.82.10.235", nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" Content-Length: 0 ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: ------------------------------------------------------------------------ ACK sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=K83mZmZH8g3Hr Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988828 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: ------------------------------------------------------------------------ INVITE sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Contact: sip:1002@209.82.10.250:1091 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE Content-Length: 187 Content-Type: application/sdp User-Agent: HelpCaster SoftPHONE Supported: com.hearme.mux Proxy-Authorization: Digest username="1002",realm="209.82.10.235",nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" d1614cd024acab0b794751285f5cb1fe",uri="sip:9999@209.82.10.235" ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: sip:9999@209.82.10.235 Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Content-Length: 0 ------------------------------------------------------------------------ send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=mHXD1FgN5SS4K Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" Content-Length: 0 ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: ------------------------------------------------------------------------ ACK sip:9999@209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1091 From: HC-Desktop <sip:1002@209.82.10.235>;tag=55409708 To: <sip:9999@209.82.10.235>;tag=mHXD1FgN5SS4K Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473(a)192.168.10.31 CSeq: 421988829 ACK Content-Length: 0 Thanks!

4 3