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Today's Topics:
1. Freeswitch session limit in freeswitch (vineet)
2. Re: Clarifications on Grandstream HT503 as FXO gateway
(Francesco Facco de Lagarda)
3. javascript scheduling multiple events excecute_on_answer
(Francesco Facco de Lagarda)
4. Re: NAT / UDP hole punching issue (Tihomir Culjaga)
---------- Forwarded message ----------
From: vineet <vineet.verma@bics.com>
To: freeswitch-users@lists.freeswitch.org
Cc:
Bcc:
Date: Wed, 28 Feb 2018 01:18:56 -0700 (MST)
Subject: [Freeswitch-users] Freeswitch session limit in freeswitch
Dears
I am experiencing that freeswitch is not able to handle more than 500
sessions even I have configured the switch.conf.xml with 5000 sessions.
Can you please help me ?
Thanks,
vineet
<settings>
--
Sent from: http://freeswitch-users.2379917.n2.nabble.com/
---------- Forwarded message ----------
From: Francesco Facco de Lagarda <francesco@delagarda.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org>
Cc:
Bcc:
Date: Wed, 28 Feb 2018 09:19:21 +0100
Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gatewayThanks kim!
.. and ofc I can set the extension to some group to make multiple phones ring, right?
But do I need to configure the HT503 as an extension?
From: FreeSWITCH-users <freeswitch-users-bounces@
lists.freeswitch.org > On Behalf Of Kim Culhan
Sent: martedì 27 febbraio 2018 22:02
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org >
Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gateway
On Mon, February 26, 2018 9:16 am, Francesco Facco de Lagarda wrote:
> Goodmorning to you all!
>
> I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6
On the Grandstream configuration web page, take a look at the Basic tab at the top.
At the bottom the Basic page there is a section:
' Unconditional Call Forward to VOIP:
The '503 will initiate a call to the extension number you specify in 'User ID'
when the FXO rings.
Hope this helps.
-kim
---------- Forwarded message ----------
From: Francesco Facco de Lagarda <francesco@delagarda.com>
To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org>
Cc:
Bcc:
Date: Wed, 28 Feb 2018 09:42:22 +0100
Subject: [Freeswitch-users] javascript scheduling multiple events excecute_on_answerI am developing a calling platform using javascript.
I have code that calculates how long THAT user is allowed to call THAT number for.
I am trying to schedule 2 events,
- 1 min before time ends, that plays a message “you have 1 minute left ..”
- The actual hangup when the time expires..
Despite a zillion tests I have been unable to schedule BOTH events..
This is my code: for simplicity’s sake I have set call time to 120 secs, with warning at 60:
if (session.ready()) {
/***
Get user, number, etc… code omitted for simplicity
**/
var sessOut = new Session("sofia/gateway/ht503/" + dialedNum + "@192.168.0.201:5062");
var totTime = 60;
sessOut.execute("set", "execute_on_answer=sched_
hangup +120 alloted_timeout") sessOut.execute("set", "execute_on_answer=sched_
broadcast +60 playback::" + soundDir + "one_min_left.wav both");
if (sessOut.ready()) {
bridge(session, sessOut);
}
sessOut.hangup();
session.hangup();
}
---------- Forwarded message ----------
From: Tihomir Culjaga <tculjaga@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Bcc:
Date: Wed, 28 Feb 2018 09:57:00 +0100
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issueRTP ports are defined. When i do a port forward for my RTP range i get my RTP audio working i guess due to rtp-auto-adjust feature on FS... but it should work without port forwarding.here simply FS is not starting to send RTP traffic to the client even if it notified its public IP:PORT in SDP on 200 OK.i see FS contacting a STUN server, getting the public IP:PORT and than ... doesn't send any RTP traffic towards the client... this is what its bugging me.T.On 28 February 2018 at 09:18, Francesco Facco de Lagarda <francesco@delagarda.com> wrote:Check your RTP ports .. in the fs config and the port forwarding on firewalls.
Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know!
Good luck!
F
From: FreeSWITCH-users <freeswitch-users-bounces@list
s.freeswitch.org > On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:01
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org >
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue
does anyone have a clue ? :=)
On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga@gmail.com> wrote:
hi,
I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :(
my setup is like this:
Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>
FreeSWITCH version: 1.6.12~64bit ( 64bit)
I have a separate profile configured for TLS:
<param name="rtp-ip" value="192.168.100.60"/>
<param name="sip-ip" value="192.168.100.60"/>
<param name="apply-nat-acl" value="rfc1918"/>
<param name="ext-sip-ip" value="stun:stun.freeswitch.or
g "/><param name="ext-rtp-ip" value="stun:stun.freeswitch.or
g "/>
<!--<param name="aggressive-nat-detection
" value="true"/>-->
<param name="tls-only" value="true"/>
<param name="tls-sip-port" value="15061"/>
============================================================ ============================== ======= Name tls-public
Domain Name N/A
Auto-NAT false
DBName sofia_reg_tls-public
Pres Hosts 192.168.100.60,192.168.100.60
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.100.60
Ext-RTP-IP stun:stun.freeswitch.org
SIP-IP 192.168.100.60
Ext-SIP-IP 85.114.41.180
TLS-BIND-URL sips:mod_sofia@85.114.41.180:1
5061;maddr=192.168.100.60;tran sport=tls WS-BIND-URL sip:mod_sofia@192.168.100.60:
5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.100.60:
7443;transport=wss HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN PCMA
CODECS OUT PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG true
PROXY-MEDIA false
ZRTP-PASSTHRU false
AGGRESSIVENAT false
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 2
FAILED-CALLS-OUT 2
REGISTRATIONS 0
i manage to register the phone with no problems but when i call the phone i get no audio;
bgapi expand originate ${sofia_contact(tls-profile/ag
ent2/nexios@192.168.100.60)} &echo()
FS sends the invite as:
SDP in INVITE message from FS
v=0
o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180
s=FreeSWITCH
c=IN IP4 85.114.41.180
t=0 0
m=audio 17480 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
SIP Client responds with:
SDP in 200 OK from the client
v=0
o=- 3728718779 3728718780 IN IP4 213.147.96.240
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 8 101
c=IN IP4 213.147.96.240
b=TIAS:64000
a=rtcp:4003 IN IP4 213.147.96.240
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
So the UDP stream is: client( 4002 ) <> ( 17480 )FS
when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client.
so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ?
in FS logs i see 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS
2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 entering state [ready][200]2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA
:8:8000:20:64000:1] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101@8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 PCMA/8000 20 ms 160 samples 64000 bits 1 channels2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Original read codec set to PCMA:82018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101@8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Set 2833 dtmf send payload to 101 recv payload to 1012018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/n
exios@213.147.96.240:10551 ] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 202018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003
2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003
2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 2
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Set 2833 dtmf send payload to 1012018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Set 2833 dtmf receive payload to 1012018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Set rtp dtmf delay to 402018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/n
exios@213.147.96.240:10551 ] has been answered2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/n
exios@213.147.96.240:10551 ) Callstate Change RINGING -> ACTIVE2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/n
exios@213.147.96.240:10551 ]2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 Flipping CID from "" <0000000000> to "Outbound Call" <nexios>2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/n
exios@213.147.96.240:10551 ) State Change CS_CONSUME_MEDIA -> CS_EXECUTE2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:58
4 (sofia/tls-public/sip:agent2/n exios@213.147.96.240:10551 ) Running State Change CS_EXECUTE2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:65
0 (sofia/tls-public/sip:agent2/n exios@213.147.96.240:10551 ) State EXECUTE2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 SOFIA EXECUTE2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:32
8 sofia/tls-public/sip:agent2/ne xios@213.147.96.240:10551 Standard EXECUTEEXECUTE sofia/tls-public/sip:agent2/ne
xios@213.147.96.240:10551 echo()
Regards,
Tihomir.
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