October 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ciphers not available for verto wss (neither wss on sip profile)
by GM phy 26 Oct '17

26 Oct '17

Hello, im having this issue and by now i found no way to figure it out. enabling tls on internal, if a exec openssl s_client -connect 10.10.10.9:5061 -tls1_2 Connection is established and server returns my certificate and this information -- New, TLSv1/SSLv3, Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-ECDSA-AES256-GCM-SHA384 ... -- When i run the same comand for port 7443 (the wss binding on internal profile) or port 8082 (verto wss) i get (same on both cases) -- New, TLSv1/SSLv3, Cipher is ECDH-RSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDH-RSA-AES256-GCM-SHA384 ... -- This causes that all of the webrtc phones we're using (some with sipjs, some with verto) are unable to establish the wss socket in order to register and make calls. On the browser console i get Protocol mismatch, and on a pcap, server rejects the ssl handshake with code 40. I've been working with the code on mod_verto.c and ws.c with no luck. basically what im looking for is for verto and/or wss on internal profile, to support Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 (or ECDHE-ECDSA) in order to webrtc can connect to fs. any help will be really appreciated. thanks in advance

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Control register behavior via a lua script when REGISTER?
by Charles Yu 26 Oct '17

26 Oct '17

Hi, Is there a way to control register behavior via a lua script when receiving sip REGISTER ? Such like: I can reject this REGISTER if it does not match some condition. Trigger a sofia::register_failure event? Any ideas?

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Adding authorization to SIP messaging
by Marko Mrvelj 25 Oct '17

25 Oct '17

Hi all, We are experimenting with chatplan, in order to get basic chat functionality. Messages do work as expected, but there is a problem that message can be submitted to server without any authorization. Could somebody tell me if it is possible to restrict messages to only authorized users? How can this be done? Thanks in advance, Marko

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recordFile not working when hangup
by Luis Daniel Lucio Quiroz 24 Oct '17

24 Oct '17

Before submitting anything, I prefer to do a public question. I have been reported in a Lua script that when people hang up, the recording is not saved in the filesystem. However, when you stop talking I am being reported that it works. The issue on FS 1.6.16 and 1.6.19. Has anyone had the same issue? -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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Pass customized SIP header in INVITE from a leg to b leg
by kaiduan xie 24 Oct '17

24 Oct '17

Hi, With default mode, for bridged call FreeSwitch strips the non-standard SIP header in INVITE from a leg, is there any configuration to let FreeSwitch pass the header as it is to b leg? Many thanks, /Kaiduan

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Freeswitch Question regarding re-use of previously negotiated codec in response to SIP reINVITE w/o SDP
by Arslan Saeed 24 Oct '17

24 Oct '17

Hello All, Seeking comments/suggestions from the community on a particular Freeswitch (FS) behavior observed. We see that if there is a SIP reINVITE message (in-dialogue) from far end switch coming to FS, it responds back with the last negotiated codec in the SDP offer as opposed to forming a brand new SDP offer with all the supported codecs listed. Now according to my understanding, It is recommended (not mandated) that the SIP entity should respond to a SIP reINVITE w/o SDP with all the Codecs it can support and not just respoind with narrowed down list of codecs (as negotiated in previous SDP offer/answer exchange). This allows the far side a way to re-negotiate the codec from scratch mid dialogue if there is a need (for example call getting transferred to new SIP UA) Any configuration option that can influence this behavior? Regards Arslan

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Transcoding on audio file with wav extension and encoded as 8 bit ulaw
by Aqs Younas 22 Oct '17

22 Oct '17

Greetings list, Just want to confirm will trancoding happen while playing a file having format as below. root@debian:/opt/tmp# soxi song.wav Input File : 'song.wav' Channels : 1 Sample Rate : 8000 Precision : 14-bit Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors File Size : 441k Bit Rate : 64.0k Sample Encoding: 8-bit u-law If i get PCMU as only codec in invite. Here .wav is just a container wrapping PCMU audio. Best Regards,

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Proxy mode, CDR and codec
by Gregor Nanger 22 Oct '17

22 Oct '17

Hi! On one FS I am using proxy mode media, just to bridge call from point A to point B. It works ok. But have problems with parsing CDR. How can I check (which variable) what codec was proposed from pont A and which codec was negotiated? Is this possible in proxy mode? Best regards, Gregor

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route question
by alex 21 Oct '17

21 Oct '17

Hello: I want to call 1001 by inputting 1001 through this route. <extension name="outgoing"> <condition field="destination_number" expression="^1001$"> <action application="bridge" data="freetdm/wp1/a/$1"/> </condition> </extension> But it didn't through the route. Why is that? What should i do? This are CLI log: 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000(a)192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000(a)192.168.1.70 to XML[1000@default] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000(a)192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000(a)192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000(a)192.168.1.70) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000(a)192.168.1.70 [CS_DESTROY]

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IVR missing start of audio files
by Rick Jarvis 21 Oct '17

21 Oct '17

I consistently have a problem where the first few seconds of IVR prompts is being missed. I tend to fudge it by putting several seconds of silence at the beginning of each wav file, but this gets to be a bit of a pain. How does everyone else handle this?

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