February 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch session limit in freeswitch
by vineet 28 Feb '18

28 Feb '18

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/

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BLF on Yealink (public "ping" to Kevin Wormington).
by Bob Smith 28 Feb '18

28 Feb '18

Hi, I've sent this to list just incase anyone else wants to chip in. I Googled up an old post from Mr Wormington (August 16 2013 @ 16:09, subject "BLF"). In it, Kevin describes how he has encountered behaviour with Yealink where the BLF remains green despite the subject of the BLF being no longer registered. There seems to have been very little in public follow up to Kevin's post at the time, so I was wondering if anyone could enlighten me as to what the solution is, because I am facing the same quandry. Thanks ! Bob

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 140, Issue 93
by Raviram Chandran 28 Feb '18

28 Feb '18

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(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > 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(a)bics.com> > To: freeswitch-users(a)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(a)delagarda.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)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(a)lists.freeswitch.org> *On > Behalf Of *Kim Culhan > *Sent:* martedì 27 febbraio 2018 22:02 > *To:* FreeSWITCH Users Help <freeswitch-users(a)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(a)delagarda.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)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(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)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(a)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(a)lists.freeswitch.org> *On >> Behalf Of *Tihomir Culjaga >> *Sent:* mercoledì 28 febbraio 2018 09:01 >> *To:* FreeSWITCH Users Help <freeswitch-users(a)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(a)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="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@85.114.41.180:15061 >> >> TLS-BIND-URL sips:mod_sofia@85.114.41.180:1 >> 5061;maddr=192.168.100.60;transport=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/agent2/ >> nexios(a)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@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/nexios@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@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@8000 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 >> sofia/tls-public/sip:agent2/nexios@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@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@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@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@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@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@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@213.147.96.240:10551] >> >> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 >> sofia/tls-public/sip:agent2/nexios@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@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@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@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@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@213.147.96.240:10551 Standard EXECUTE >> >> EXECUTE sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 echo() >> >> >> >> >> >> >> >> Regards, >> >> Tihomir. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _______________________________________________ > 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 > >

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Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gateway
by Kim Culhan 28 Feb '18

28 Feb '18

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

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Receiving audio despite SDP with a=sendonly
by Larry Hemenway 27 Feb '18

27 Feb '18

Hello, Is there a way to establish a call with one-way audio on a call from the start? I'm currently sending a request with the following SDP - (note a=sendonly): v=0 o=Larry 2890844526 2890844526 IN IP4 127.0.0.1 s= My Session c=IN IP4 172.22.112.1 t=0 0 m=audio 49170 RTP/AVP 0 8 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 FreeSWITCH is sending back the right response (note the a=recvonly): v=0 o=FreeSWITCH 1519171720 1519171721 IN IP4 172.17.0.2 s=FreeSWITCH c=IN IP4 172.17.0.2 t=0 0 m=audio 21768 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=ptime:20 But I'm still receiving audio from FreeSWITCH. It is only after I resend the SDP via a reinvite that the audio stops being sent from FreeSWITCH. Larry

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Lua question (re: FollowMe Dialplan Example from the docs)
by Bob Smith 27 Feb '18

27 Feb '18

Hello, The docs (https://freeswitch.org/confluence/display/FREESWITCH/Dialplan+FollowMe) have an interesting example, the snappily titled "FollowMe Dialplan Example 6". This looks interesting. But I have one concern, and being a Lua newbie, I don't know how to resolve it. My reading of the script is that it does not take care of the somewhat obvious problem of what happens if the cell phone goes to voicemail ? My understanding of the script as presented is that it would just sit there forever waiting for a reply ? Could a resident Lua guru kindly enlighten this newbie as to how I could timeout the session ? Thanks ! Bob

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Passthrough conference video frame rate
by Muhammad Umair 27 Feb '18

27 Feb '18

Hi, I am wondering what is the video frame rate in conference passthrough mode. Any ideas?

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Add user to conference from verto communicator
by tudorgabriell Tudor 27 Feb '18

27 Feb '18

Hello, Please tell me if there is a way to add a new participant to conference from verto communicator? I would like to dial a number from verto communicator after a conference has started to add this as a new participant to the conference. Thank you, Tudor

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bind_local is not working for xmd_cdr.conf
by vineet 27 Feb '18

27 Feb '18

Dears, I am trying to POST the xml cdr generated by free which to a particular url defined in “xmd_cdr.conf” But my http request is getting blocked on firewall. because the source IP used in while sending the request is not correct. I tried to change the source IP by modifying the following line in xml_curl.conf.xml But it did not helped. Can some please help me to change the source IP while sending the http POST for xml_cdr? Thanks, Vineet -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Test email. No email received after 23-Feb.
by Uday kumar 27 Feb '18

27 Feb '18

Testing email. No email received after 24-Feb. -- Thanks & Regard Uday. Mobile:- +91-9377579349

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