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Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
by Kamrul Khan 20 Aug '14

20 Aug '14

Hi,Thanks for your reply. In that case, how to know what exactly the issue is. The requested codecs in my INVITE is as follows:a=rtpmap:109 opus/48000/2a=ptime:20a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ hosta=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ hosta=rtcp-muxThe codec installed in my freeswitch:show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc Is is rejecting just because I dont have opus installed in my freeswitch? But, I do have ulaw and alaw. The same configuration works fine with asterisk and I dont have OPUS installed there aswell.Asterisk codec list: ID TYPE NAME DESCRIPTION -------------------------------------------------------------------------------- --- 100001 audio g723 (G.723.1) 100002 audio gsm (GSM) 100003 audio ulaw (G.711 u-law) 100004 audio alaw (G.711 A-law) 100011 audio g726 (G.726 RFC3551) 100006 audio adpcm (ADPCM) 100019 audio slin (16 bit Signed Linear PCM) 100007 audio lpc10 (LPC10) 100008 audio g729 (G.729A) 100009 audio speex (SpeeX) 100016 audio speex16 (SpeeX 16khz) 100010 audio ilbc (iLBC) 100005 audio g726aal2 (G.726 AAL2) 100012 audio g722 (G722) 100021 audio slin16 (16 bit Signed Linear PCM (16kHz)) 300001 image jpeg (JPEG image) 300002 image png (PNG image) 200001 video h261 (H.261 Video) 200002 video h263 (H.263 Video) 200003 video h263p (H.263+ Video) 200004 video h264 (H.264 Video) 200005 video mpeg4 (MPEG4 Video) 400001 text red (T.140 Realtime Text with redundancy) 400002 text t140 (Passthrough T.140 Realtime Text) 100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 100017 audio testlaw (G.711 test-law) 100015 audio g719 (ITU G.719) 100028 audio speex32 (SpeeX 32khz) 100020 audio slin12 (16 bit Signed Linear PCM (12kHz)) 100022 audio slin24 (16 bit Signed Linear PCM (24kHz)) 100023 audio slin32 (16 bit Signed Linear PCM (32kHz)) 100024 audio slin44 (16 bit Signed Linear PCM (44kHz)) 100025 audio slin48 (16 bit Signed Linear PCM (48kHz)) 100026 audio slin96 (16 bit Signed Linear PCM (96kHz)) 100027 audio slin192 (16 bit Signed Linear PCM (192kHz)) 100018 audio silk8 (SILK Custom Format 8khz) 100018 audio silk12 (SILK Custom Format 12khz) 100018 audio silk16 (SILK Custom Format 16khz) 100018 audio silk24 (SILK Custom Format 24khz) --Forwarded Message Attachment-- From: sertys(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 12:43:11 +0200 Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488 He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call. On Wed, Aug 20, 2014 at 9:01 AM, <dodul(a)live.com> wrote: Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 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..." _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 --Forwarded Message Attachment-- From: matt(a)inveroak.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 11:59:35 +0100 Subject: Re: [Freeswitch-users] Internal IP showing in SDP Hi Brian, it seems that this was not the issue. Some more details: We have an opensips server using the load_balancer module to route the calls to the freeswitch box. (https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) When I route the calls directly to the freeswitch's external IP, the calls connect fine and have audio. However when I route the calls to the opensips server first, the call will connect but without audio. The strange thing is Opensips uses Freeswitch's internal address (from the load_balancer table) to forward the INVITE packet, yet when the ACK packet comes in from the provider opensips tries to forward it to the FS external IP address rather than the internal IP. I understand this is probably more an opensips question, so will understand if I should post this to their mailing list, but any tips would be appreciated. thanks Matt On 19/08/2014 17:25, Matt Broad wrote: thanks brian, will give it a go and see what happens. thanks Matt On 19/08/2014 17:06, Brian West wrote: Can you try prefixing the ext-rtp-ip and ext-sip-ip with autonat:0.0.0.0 and make sure local-network-acl is set to rfc1918.auto? On Tue, Aug 19, 2014 at 9:20 AM, Matt Broad <matt(a)inveroak.com> wrote: Hi, does it make any difference if my internal IP address is being sent in the SDP connection information? I have 2 sip providers where 1 I can bridge calls and receive audio and 1 that I can bridge calls but get no audio. Even with the below settings the SDP message body contains c=IN IP4 192.168.1.1. I have vars set as: <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.ip.goes.here"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.ip.goes.here"/> I have setup the external profile as follows: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> running sofia status profile external gives (replace 0.0.0.0 with the real external IP declared in vars): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.1 Ext-RTP-IP 0.0.0.0 SIP-IP 192.168.1.1 Ext-SIP-IP 0.0.0.0 URL sip:mod_sofia@0.0.0.0:5080 BIND-URL sip:mod_sofia@0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp WS-BIND-URL sip:mod_sofia@192.168.1.1:5086;transport=ws HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 8 FAILED-CALLS-IN 1 CALLS-OUT 8 FAILED-CALLS-OUT 1 REGISTRATIONS 0 thanks Matt _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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

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Re: [Freeswitch-users] Installing OPUS codec
by Kamrul Khan 20 Aug '14

20 Aug '14

I installed from source as I mentioned mod_opus is in my source directory. But the mod_opus.so is not in my freeswitch installation module directory. So, for some reason the module didnt get build during my installation. From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 134 To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 19:54:30 +0400 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..." --Forwarded Message Attachment-- From: brian(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:47:48 -0500 Subject: Re: [Freeswitch-users] Installing OPUS codec mod_opus is part of the default build, are you installing via packages or from source? On Tue, Aug 19, 2014 at 7:57 PM, Kamrul Khan <dodul(a)live.com> wrote: Hi, Can anyone tell me how to install OPUS module in FREESWITCH ? I have the source in my source directory. _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: brian(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:49:38 -0500 Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488 Show the logs on FreeSWITCH from right before the inbound invite arrives and after the call is rejected with 488, If you're doing WebRTC you have no reason to even be off on the SRTP wiki pages in the first place. On Wed, Aug 20, 2014 at 5:43 AM, Daniel Ivanov <sertys(a)gmail.com> wrote: He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call. On Wed, Aug 20, 2014 at 9:01 AM, <dodul(a)live.com> wrote: Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 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..." _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: andrew(a)cassidywebservices.co.uk To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 16:53:58 +0100 Subject: Re: [Freeswitch-users] Freeswitch dialplan issue secondly you may also need to set the actions as inline to use a variable you've just set. add inline="true" to you actions. On 20 August 2014 15:03, Steven Ayre <steveayre(a)gmail.com> wrote: Is the first condition matching? On 20 August 2014 13:20, Nick Zaitsev <nick.zaitsev(a)mail.ru> wrote: Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${ext}(a)10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 -- Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director T 03300 100 960 F 03300 100 961 E andrew(a)cassidywebservices.co.uk W www.cassidywebservices.co.uk

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Fw: Freeswitch dialplan issue
by Nick Zaitsev 20 Aug '14

20 Aug '14

Yes,the first condition is mathcing. From: Steven Ayre Sent: Wednesday, August 20, 2014 6:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch dialplan issue Is the first condition matching? On 20 August 2014 13:20, Nick Zaitsev <nick.zaitsev(a)mail.ru> wrote: Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${mailto:ext%7D@10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev _________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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

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Freeswitch dialplan issue
by Nick Zaitsev 20 Aug '14

20 Aug '14

Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${ext}(a)10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev

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Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
by dodul＠live.com 20 Aug '14

20 Aug '14

Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 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..."

3 2

Authenticating a user through ESL script
by sami sakly 20 Aug '14

20 Aug '14

Hi, I'm looking to issue calls from the CLI (using originate command ) the same way it's done through any Dialer like SIPml5. I think I need to do something about the dialplan because once I'm calling using SIPml5 I'm passing "through" default dialplan and when I'm only using originate from external script I'm passing "through" Public dialplan. May be something about authenticated user ? 2014-08-18 18:20:37.712649 [INFO] mod_dialplan_xml.c:558 Processing <0000000000>->1000 in context public for the originate command I'm using esl.php : $ec1 = $sock->bgapi("originate","{origination_uuid=$uuid1}sofia/internal/ 1000(a)12.0.0.104 &park()"); Thanks for your help. Regards,

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mod_spidermonkey / freeswitch-v8.x86_64
by Mujahid Ali 20 Aug '14

20 Aug '14

when i install freeswitch-v8.x86_64 by yum like yum install freeswitch-v8.x86_64 it install but there is no file mod_v8.so in /usr/lib64/freeswitch/mod/ even else every thing is OK is it completely install or not ?

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Registering with digest authentication, but empty password. How?
by Bob Hartwig 20 Aug '14

20 Aug '14

Hi all, This is going to sound like a weird problem. I'm trying to test a SIP trunk from a vendor that requires registering with digest authentication, but an empty password, and I don't know how to specify the empty password in my profile. If I configure my sofia profile like this "<param name="password" value="boguspassword"/>", the vendor gives me a 407, so I know that they need that empty password. If I configure it like this: "<param name="password" value=""/>", Freeswitch complains that "[ERR] sofia.c:2703 ERROR: password param is REQUIRED!". So is there any way to configure the profile with an empty password? Thanks, Bob Hartwig

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Freeswitch srtp rejecting SRTP with 488
by Kamrul Khan 19 Aug '14

19 Aug '14

Hi, Below is my dialplan: <extension name="cext"> <condition field="destination_number" expression="^(2000)$"> <action application="set" data="rtp_secure_media=true"/> <action application="bridge" data="user/bob"/> </condition> <condition field="destination_number" expression="^(3000)$"> <action application="set" data="rtp_secure_media=true"/> <action application="bridge" data="user/bob"/> </condition> </extension></context> You can see I have rtp_secure_media=true for both extension. However, when I try to call I get rejected with 488 Not acceptable here. In my freeswitch console log I find sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION. What should I do to fix this? Below is my complete console log: 2014-07-22 19:51:32.572499 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-07-22 19:51:32.707332 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-07-22 19:51:32.724263 [NOTICE] switch_channel.c:669 New Channel sofia/internal/alice(a)192.168.146.133 [4101d950-1214-11e4-933c-61443ed3c9c7]2014-07-22 19:51:32.725481 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_NEW2014-07-22 19:51:32.725481 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/alice(a)192.168.146.133) State NEW2014-07-22 19:51:32.762327 [DEBUG] sofia.c:4153 Channel sofia/internal/alice(a)192.168.146.133 entering state [received][100]2014-07-22 19:51:32.762327 [DEBUG] sofia.c:4164 Remote SDP:v=0o=Mozilla-SIPUA-31.0 12934 1 IN IP4 0.0.0.0s=Doubango Telecom - firefoxt=0 0a=ice-ufrag:5e3a0151a=ice-pwd:9df6e8f0963f033e8573a23e4ce62298a=fingerprint:sha-256 90:BD:D7:9C:DE:66:5D:86:D4:8A:61:89:75:07:EF:F9:9A:06:AE:29:C8:1D:BD:C3:BD:C0:41:12:01:B5:45:FDm=audio 56772 UDP/TLS/RTP/SAVPF 109 0 8 101c=IN IP4 184.69.59.132a=rtpmap:109 opus/48000/2a=ptime:20a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ hosta=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ hosta=rtcp-mux 2014-07-22 19:51:32.762327 [NOTICE] sofia.c:4354 Hangup sofia/internal/alice(a)192.168.146.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]2014-07-22 19:51:32.762327 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/alice(a)192.168.146.133 [KILL]2014-07-22 19:51:32.762327 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.762327 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_HANGUP2014-07-22 19:51:32.762327 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP2014-07-22 19:51:32.762327 [DEBUG] mod_sofia.c:414 Channel sofia/internal/alice(a)192.168.146.133 hanging up, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 4882014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:46 sofia/internal/alice(a)192.168.146.133 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP going to sleep2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/alice(a)192.168.146.133) State Change CS_HANGUP -> CS_REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING going to sleep2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/alice(a)192.168.146.133) State Change CS_REPORTING -> CS_DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-07-22 19:51:32.780344 [DEBUG] switch_core_session.c:1164 Session 138 (sofia/internal/alice(a)192.168.146.133) Locked, Waiting on external entities2014-07-22 19:51:32.780344 [NOTICE] switch_core_session.c:1182 Session 138 (sofia/internal/alice(a)192.168.146.133) Ended2014-07-22 19:51:32.780344 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/alice(a)192.168.146.133 [CS_DESTROY]2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY2014-07-22 19:51:32.780344 [DEBUG] mod_sofia.c:341 sofia/internal/alice(a)192.168.146.133 SOFIA DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:60 sofia/internal/alice(a)192.168.146.133 Standard DESTROY2014-07-22 19:51:32.780344 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY going to sleep

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[ANN] openerp-freeswitch-connector
by Trever L. Adams 19 Aug '14

19 Aug '14

Hello everyone, I am announcing this here in hopes that some of you may find this useful. I have forked http://www.akretion.com/open-source-contributions/openerp-asterisk-voip-con… and made it work with FreeSWITCH. At the moment it works for me. The only feature currently working is click to dial under customers. I will be implementing all of the features found in the original over the coming weeks. I will also be adding some that they do not have. Anyone, of course, is free to use this under the terms of the license. Anyone wishing to help out with documentation, code, features, what not, you are also welcome. The repository can be found at https://github.com/treveradams/openerp-freeswitch-connector Thank you, Trever

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