- FreeSWITCH-users - FreeSWITCH Mailing Lists

Custom Headers with mod_callcenter
by Ari Sonesh 08 Mar '18

08 Mar '18

Can anyone advise when using the mod callcenter ... how to pass custom sip headers from the calling party call leg to the agent call leg? thanks! *Ari Sonesh *

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NAT / UDP hole punching issue
by Tihomir Culjaga 07 Mar '18

07 Mar '18

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@85.114.41.180:15061 ;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/ nexios@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.

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Media storage for Freeswitch
by Melek Oktay 06 Mar '18

06 Mar '18

Hi, I need a solution to store and read media files from Freeswitch servers (I have multiple Freeswitch). Media files are recorded voicemails, recorded calls and user files like welcome sounds, IVR sounds, waiting music, etc. Currently this is solved by using a mount to a SMB share (NFS) on local storage. This works good but I think it has a drawbacks like it is not scalable.. I wonder, what kind of solution exist in the literature ? What are the best practices that you face? Best Regards, Melek P.S. At first a solution with a distributed file system was tried with GlusterFS. This had some drawbacks, especially with high latency synchronization between nodes. This approach was dropped.

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Internal Sip profile error
by Social Boh 06 Mar '18

06 Mar '18

Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: *freeswitch -nonat -nf* IPv4 internal profile working fine: * sofia status* Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) 45.77.161.19 alias internal ALIASED external profile sip:mod_sofia@45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) internal profile sip:mod_sofia@45.77.161.19:5060 RUNNING (0) ================================================================================================= If I use script daemon *systemctl start freeswitch* with this parameters: *-nonat -nf -nc* I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia@45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) external profile sip:mod_sofia@45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe

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[mod_conference] playback .wav after x seconds of silence
by Mititelu Stefan 06 Mar '18

06 Mar '18

Hi, Is there a way to playback a random .wav message after silence threshold has been kept for more than X seconds, in a conference? Thank you, Stefan

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pgsql core and odbc
by Madovsky 06 Mar '18

06 Mar '18

Hi, I noticed that if I use pgsql core protocol in switch.conf.xml (pgsql://) to a remote postgresql through VPN ps shows an idle connection taking between 15 and 50% of cpu. with odbc there is no such kind of issue. jira or a special setting needed? thanks

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(no subject)
by Ali Haider 05 Mar '18

05 Mar '18

[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] Usage: fs_cli [-H <host>] [-P <port>] [-p <se how you can ressolve this error kindly tell me please Sent from my iPhone

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Re: [Freeswitch-users] (no subject)
by Raimundo Pérez Nieves 05 Mar '18

05 Mar '18

Try this: Go to path /usr/bin and execute ./freeswitch It starts FS in secure mode, and you can debug the error. In my case, it was bad close tag in html and I just shutdown FS with that error, so It did not open. Doing this gave the idea of the problem. Hope solves Enviado desde mi iPhone El 05/03/2018, a la(s) 17:52, Ali Haider <alihaider.4189(a)gmail.com <mailto:alihaider.4189@gmail.com>> escribió: > > [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] > Usage: fs_cli [-H <host>] [-P <port>] [-p <se > how you can ressolve this error kindly tell me please > Sent from my iPhone > >

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ClueCon Weekly with Matrix!
by Kathleen King 05 Mar '18

05 Mar '18

Hello, Don't miss the call this Wednesday with Matthew Hodgson from Matrix! He will be talking about everything new they are working on and some details of the exciting new release. You can join the call by dialing 888 at https://conference.freeswitch.org/vc/ or watch it live at https://youtu.be/PIDKy2JtiJc Kathleen King | Public Relations / Administrative Assistant FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: Kathleen(a)freeswitch.com Mobile: 703-859-3757 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwg…>

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IVR in Freeswitch using LUA
by Vishal Pai 05 Mar '18

05 Mar '18

Hi All I have built a IVR system in Freeswitch using LUA. In my IVR i am doing basic operation like select, insert etc. with remote MSSQL database. I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit (git b1b21d0 2017-11-20 16:34:16Z 64bit) My concern is with the CPU usage it get more than 100% in just 3 calls. Please suggest what should i do to get normal usage of CPU.

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