[Freeswitch-users] FreeSWITCH-users Digest, Vol 140, Issue 93
Raviram Chandran
chandranraviram at gmail.com
Wed Feb 28 09:25:31 UTC 2018
Hi Vineet,
We got our telephony solutions developed by an Indian company (StarTele
Logic) and its working very well, we are running more then 1000 concurrent
calls. I am not sure how they developed that but you should talk them..may
be they can help you out.
All the best.
Ram.
On Wed, Feb 28, 2018 at 2:27 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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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 at bics.com>
> To: freeswitch-users at 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>
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> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
>
>
>
> ---------- Forwarded message ----------
> From: Francesco Facco de Lagarda <francesco at delagarda.com>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:19:21 +0100
> Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO
> gateway
>
> Thanks 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 at lists.freeswitch.org> *On
> Behalf Of *Kim Culhan
> *Sent:* martedì 27 febbraio 2018 22:02
> *To:* FreeSWITCH Users Help <freeswitch-users at 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 at delagarda.com>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:42:22 +0100
> Subject: [Freeswitch-users] javascript scheduling multiple events
> excecute_on_answer
>
> I 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. 1 min before time ends, that plays a message “you have 1 minute
> left ..”
> 2. 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 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:57:00 +0100
> Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue
> RTP 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 at 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 at lists.freeswitch.org> *On
>> Behalf Of *Tihomir Culjaga
>> *Sent:* mercoledì 28 febbraio 2018 09:01
>> *To:* FreeSWITCH Users Help <freeswitch-users at 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 at 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.org"/>
>>
>> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>>
>>
>>
>> <!--<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-URL sip:mod_sofia at 85.114.41.180:15061
>>
>> TLS-BIND-URL sips:mod_sofia at 85.114.41.180:1
>> 5061;maddr=192.168.100.60;transport=tls
>>
>> WS-BIND-URL sip:mod_sofia at 192.168.100.60:5066;transport=ws
>>
>> WSS-BIND-URL sips:mod_sofia at 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/agent2/
>> nexios at 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/nexios at 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 at 8000
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms
>> 160 samples 64000 bits 1 channels
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read
>> codec set to PCMA:8
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
>> telephone-event payload to 101 at 8000
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> send payload to 101 recv payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
>> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
>> port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
>>
>> 2018-02-27 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/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> send payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> receive payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf
>> delay to 40
>>
>> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
>> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been
>> answered
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate
>> Change RINGING -> ACTIVE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
>> Resulted in Success: [sofia/tls-public/sip:agent2/n
>> exios at 213.147.96.240:10551]
>>
>> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
>> sofia/tls-public/sip:agent2/nexios at 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/nexios at 213.147.96.240:10551) State Change
>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
>> Change CS_EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
>>
>> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()
>>
>>
>>
>>
>>
>>
>>
>> Regards,
>>
>> Tihomir.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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