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SRTP switch_core_media.c:4522 Crypto not negotiated but required.
by Miller, Michael 17 Aug '17

17 Aug '17

I'm new to Freeswitch and have udp, tcp and tls only calls working. However; I am having problems with tls and srtp. Call flow is bria client -- (udp+rdp) --> freeswitch -- (tls+srtp) --> sbc In my invite, I send the following SDP from Freeswitcch: m=audio 20606 RTP/SAVP 9 8 0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:6dYwo00ZMUlaJXkdnTWAVsaqBsxXvVnPBO14sbO7 a=ptime:20 In the 180 and 200 response, I get: m=audio 6206 RTP/SAVP 8 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:MrKNE3L5SSc9yb1o7G5BWJkcpLFvprrctnH/c6go a=rtpmap:8 PCMA/8000 a=sendrecv a=maxptime:20 Everything looks good. After the 200 ok, I see the following in the freeswitch log: 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:1395 looking for crypto suite [AES_CM_128_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:MrKNE3L5SSc9yb1o7G5BWJkcpLFvprrctnH/c6go] 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:1400 Found suite AES_CM_128_HMAC_SHA1_80 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:1466 Set Remote Key [5 AES_CM_128_HMAC_SHA1_80 inline:MrKNE3L5SSc9yb1o7G5BWJkcpLFvprrctnH/c6go] 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-08-17 08:48:34.801061 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] After receiving the ACK, I see the following in the fs log: 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:120:48000:20:0:2]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:120:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:120:48000:20:0:2]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:122:16000:20:0:1]/[G722:9:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:122:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:122:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101@8000 2017-08-17 08:48:34.821064 [WARNING] switch_core_media.c:4522 Crypto not negotiated but required. 2017-08-17 08:48:34.821064 [DEBUG] switch_core_media.c:4767 sofia/internal/7202007917(a)10.251.145.154 Set 2833 dtmf send payload to 101 recv payload to 101 2017-08-17 08:48:34.821064 [NOTICE] switch_channel.c:3812 Hangup sofia/internal/7202007917(a)10.251.145.154 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2017-08-17 08:48:34.821064 [DEBUG] switch_ivr_originate.c:3661 sofia/internal/7202007917(a)10.251.145.154 Media Establishment Failed. My dial-plan looks like: <include> <extension name="CBRNBS703_TLS_7202007909"> <condition field="destination_number" expression="7202007909"\> <action application="set" data="dtmf_type=rfc2833"/> <action application="export" data="rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80"/> <action application="export" data="sdp_secure_savp_only=true"/> <action application="bridge" data="sofia/gateway/cbrnbs703-tls/7202007909@4.4.0.181:5061;transport=tls"/> </condition> </extension> </include> Any help would be appreciated.

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Re: [Freeswitch-users] AMR Codec for A-leg and GSM Codec for B-leg
by Ron Menez 17 Aug '17

17 Aug '17

Hi, We also have same issue with AMR codec. Does anyone knows how to do this? There is no documents available regarding the AMR codec as passthrough mode. BR, Bob > On Aug 15, 2017, at 6:30 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. AMR Codec for A-leg and GSM Codec for B-leg (Ron) > 2. Re: Answer a phone remotely via API (Peter Steinbach) > > From: Ron <ron.menez(a)entropysolution.com> > Subject: [Freeswitch-users] AMR Codec for A-leg and GSM Codec for B-leg > Date: August 15, 2017 at 2:58:08 PM GMT+8 > To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> > > > Hi All, > > Is it possible to use AMR Codec to A-leg and GSM Codec for B-leg or vice versa? > > Tried the following configuration in sip profiles: > > <param name="inherit_codec" value="true"/> > <param name="inbound-late-negotiation" value="true”/> > <param name="disable-transcoding" value="false”/> > > We received an error in the logs: > > e178024a-8180-11e7-9d17-e38b940a41a7 2017-08-15 06:13:45.955687 [DEBUG] switch_ivr_originate.c:3833 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > e178024a-8180-11e7-9d17-e38b940a41a7 2017-08-15 06:13:45.955687 [INFO] mod_dptools.c:3418 Originate Failed. Cause: INCOMPATIBLE_DESTINATION > e178024a-8180-11e7-9d17-e38b940a41a7 2017-08-15 06:13:45.955687 [NOTICE] switch_channel.c:4822 Hangup sofia/internal/09589999999(a)10.1.0.5 <mailto:sofia/internal/09589999999@10.1.0.5> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > We also tried to use the absolute_codec_string configuration in dialplan: > > <action application="bridge" data="{absolute_codec_string=GSM}sofia/internal/${destination_number}@10.1.0.4:5080"/> > > The error we received on this is: > > 2017-08-15 06:17:29.755700 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode! > 2017-08-15 06:17:29.755700 [ERR] switch_core_io.c:1434 Codec AMR encoder error! > > For your kind advice please. > > TIA. > > Best Regard, > > Ron Menez > ron.menez(a)entropysolution.com <mailto:ron.menez@entropysolution.com> > > > > > > > > From: Peter Steinbach <lists(a)telefaks.de> > Subject: Re: [Freeswitch-users] Answer a phone remotely via API > Date: August 15, 2017 at 6:30:19 PM GMT+8 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Thanks Brian, > > that worked for me. > > uuid_phone_event <uuid> talk > > on the Freeswitch CLI answered the tested phones (Yealink, Sangoma and Grandstream). > > Best regards > Peter > > On 08/14/17 22:02, Brian West wrote: >> depends on phone, see uuid_phone_event >> >> On Mon, Aug 14, 2017 at 2:36 PM, Peter Steinbach <lists(a)telefaks.de <mailto:lists@telefaks.de>> wrote: >> Hello, >> >> does there exist a command to answer a phone remotely? E.g. send a >> command via ESL to Freeswitch and then to the Phone? >> I am not looking for some kind of autoanswer. I just want to let the >> phone ring, then send some kind of event to the phone and then the phone >> answers. >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists <mailto:lists> (att) telefaks.de <http://telefaks.de/> >> Internet: www.telefaks.de <http://www.telefaks.de/> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> >> >> -- >> Brian West >> brian(a)freeswitch.org <mailto:brian@freeswitch.org> >> Twitter: @FreeSWITCH , @briankwest >> >> http://www.freeswitchbook.com <http://www.freeswitchbook.com/> >> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> >> Book a phone call (CST) <https://freeswitch.com/appointment> >> >> Allison prompts for FreeSWITCH: >> >> https://www.gofundme.com/allison-prompts-for-freeswitch <https://www.gofundme.com/allison-prompts-for-freeswitch> >> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> >> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >> Skype:briankwest >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists <mailto:lists> (att) telefaks.de > Internet: www.telefaks.de <http://www.telefaks.de/> > > > > _______________________________________________ > 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] Dockerfile for FreeSWITCH
by Robert Mundkowsky 16 Aug '17

16 Aug '17

Are there any performance issues with using Freeswitch on Docker? Thought there were issues with the Docker network stack or setup? Or is this meant to be used for small use cases? -------- Original message -------- From: David Villasmil <david.villasmil.work(a)gmail.com> Date: 8/16/17 11:32 AM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Dockerfile for FreeSWITCH Https://Github.com/davidcsi/docker-freeswitch from 2016:) On Aug 16, 2017 00:51, "Gavin Henry" <gavin.henry(a)gmail.com> wrote: Excellent work! On Mon, 6 Mar 2017 at 18:54, Brian West <brian(a)freeswitch.org> wrote: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docker If anyone wishes to submit pull requests to improve these please do so at freeswitch.org/jira I would like to work with the community to improve and document the use cases correct. Thanks, -- Brian West brian(a)freeswitch.orgTwitter: @FreeSWITCH , @briankwesthttp://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH:https://www.gofundme.com/allison-prompts-for-freeswitch Wish to schedule a meeting?http://app.timebridge.com/#/meet/freeswitchGot Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) 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-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-- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk _________________________________________________________________________ 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

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Does this configuration make sense?
by Michael Avers 16 Aug '17

16 Aug '17

I'm trying to ditch multiple registrations and go with a concept where a user (say identified internally as extension # 3005) can register any number of devices/softphones using unique SIP credentials for each one, and so extension would end up having these extensions configured: 3005@domain - primary extension, but no one registers it directly. It is used for presence and its dial_string will define what other extensions are supposed to ring 3005_desktop@domain 3005_mobile1@domain 3005_yealink@domain and so 3005's dial_string can define that the Yealink and Mobile1 extensions will ring for the first 10 seconds, and only then 3005_desktop starts to ring as well. They will all have 3005's VM box as their voicemail box. When they call out they will set presence_id to be 3005@domain and also presence_id will be set when they are called via 3005's dial_string. Anybody ever did this kind of set up? The whole idea is to have the user control which of their devices ring, if at all. Because I see more and more users registering on 3+ devices and they don't always want them to all to ring right away. Mike

6 8

Empty Re-Invite after a=inactive
by Andrew Cassidy 15 Aug '17

15 Aug '17

Hi Guys, Just wanted to check a behaviour with you guys before rushing to upgrade FreeSWITCH from 1.4.26-37 We are reciveing a reinvite with a=inactive followed by an reinvite with no SDP. I am being told the one with no SDP should revert the media the the previously established. FreeSWITCH isn't currently doing that, (or passing the empty one back to the endpoint) it's just keeping the call held. Which is the correct behaviour? Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

5 11

Channel variables on a-leg and b-leg
by Paul Mateer 15 Aug '17

15 Aug '17

Sorry. I've had to switch to a different mail account because each time I try to reply I get a "you must be a member to post to this list" message, even though I'm already subscribed. Anyway, I don't have an issue with the use of the a-leg uuid. I think my problem stems from an assumption that the ids would be the same on the client and the server. I have a client UA which is placing a call through the server, which in turn is recording the conversation. The client needs a unique way of identifying the filename used in the recording and the use of the uuid seemed like a suitable way of identifying the file. However I've realised that the call Id returned as a response to the pa call issued is a client-side id and doesn't correlate with the call id used at the server end of things. Is there any way for the client to know the call used by the server, or is there a way for the client to provide a custom id that will be accessible to the server when processing the dialplan? > The answer is all of these uuid’s are being used by the server, 1 for a leg, and one for each b leg. Why do you have an issue with using the a leg uuid? >* On Aug 14, 2017, at 1:08 PM, Paul Mateer <paul.mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: *> >* Thanks Vladyslav. *> >* After reading your reply I found relevant information under the originating_leg_uuid section of the Channel Variables confluence page. *> >* I tried modifying the Dialplan entry on the server to include both ${uuid} and $(call_uuid} in the name of the session recording file as follows *> >* <extension name="Test Dialplan entry"> *>* <condition field="destination_number" expression="^(4008)$"> *>* <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/${strftime(%Y.%m.%d_%H.%M.%S)}_${uuid}_.${call_uuid}._${caller_id_number}.wav"/> *>* <action application="export" data="dialed_extension=$1"/> *>* <action application="set" data="call_timeout=30"/> *>* <action application="set" data="hangup_after_bridge=true"/> *>* <action application="set" data="continue_on_fail=true"/> *>* <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> *>* <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> *>* <action application="bridge" data="user/$1@${domain_name}"/> *>* <action application="answer"/> *>* </condition> *>* </extension> *> >* but the name generated had the same Guid for both uuid and call_uuid (which suggests that it's the a-leg). *> > >* My client issues a call to a number using embedded FreeSWITCH as follows: *> >* FreeSWITCH.Native.Api.Execute("pa", "call sofia/gateway/<gateway>/<number> XML <caller Id> <caller name>") *> >* and gets back a response like *> >* SUCCESS:<call number>:<Guid> *> >* The Guid in the response looks like it should be a call Id because it'd the same Guid that appears in the Unique-ID headers of the FreeSWITCH events that the client receives - it just doesn't seem to match the Guid(s) used by the server when it'd generating the recording name. Is there any way for the client FreeSWITCH to know or gain access to the IDs being used by the FreeSWITCH server? *> >* Paul *> >* From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> on behalf of Vladyslav Zakhozhai <v.zakhozhai at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* Sent: 14 August 2017 16:41:13 *>* To: FreeSWITCH Users Help *>* Subject: Re: [Freeswitch-users] Channel variables on a-leg and b-leg *> >* Hi Paul, *> >* ${uuid} - is Unique Id of channel. For a-leg and b-leg it will be different (I mean that uuid is not Call-ID). You can try to use call_uuid instead. But I'm not sure about that. *> >* I think that you can use set instead of export and fetch recording data from a-leg. I'm using such approach and do not experience any problems with that. *> >* On Mon, Aug 14, 2017 at 5:01 PM, Paul Mateer <Paul.Mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:Paul.Mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>> wrote: *>* Guys, I'm using the C# interface to FreeSWITCH to add the ability to make and receive calls in an existing software application. *> >* I want to record the conversations on the server, but I need to be able to access the recording after the call is complete. *>* The Call-ID value seemed the perfect solution to this and I added the following action to my dialplan entry. *> >* <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/${strftime(%Y.%m.%d_%H.%M.%S)}_${uuid}_${caller_id_number}.wav"/> *> >* This seemed to work fine, as I would get recording files created with a Guid in the name, but on closer inspection I found that the Guid used in the filename was not the same Guid associated with the successfully placed call in the application. I assume that this is because the application sees the Call-ID for the a-leg, whereas the dialplan entry is executed in the b-leg and therefore has it's own separate Guid. *> >* Is there any way in which I can expose the Call-ID (or some other custom channel variable) for the a-leg to the b-leg so that it will be accessible within the XML dialplan and therefore be usable when creating the recording file? *> >* Thanks for any suggestions/assistance offered. *> > > >* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/ <http://www.freeswitchsolutions.com/>> *> >* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/ <http://confluence.freeswitch.org/>> *>* http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/ <http://www.cluecon.com/>> *> >* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> <http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>> *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *> > > >* -- *>* С уважением, *>* Владислав Захожай *> >* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> *> >* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org/> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> *>* http://www.cluecon.com <http://www.cluecon.com/> *> >* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> *>* http://www.freeswitch.org <http://www.freeswitch.org/> *

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rad_auth failing under load
by Joseph Waite 15 Aug '17

15 Aug '17

Hi Guys I am having a strange issue, when I have more than about 450 concurrent calls, rad_auth seems to fail to authorise calls. I see the Radius request received and replied with RAD-ACCEPT from the Radius server but I get the following error 2017-08-07 18:29:53.202133 [ERR] mod_rad_auth.c:504 RADIUS Authentication failure (RC=1) 2017-08-07 18:29:53.202133 [ERR] mod_rad_auth.c:702 An error occured during radius authorization. I have had a look as ulimit and seems to be fine. core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 128716 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 65536 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) 128716 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Anyone have any ideas? Regards

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Answer a phone remotely via API
by Peter Steinbach 15 Aug '17

15 Aug '17

Hello, does there exist a command to answer a phone remotely? E.g. send a command via ESL to Freeswitch and then to the Phone? I am not looking for some kind of autoanswer. I just want to let the phone ring, then send some kind of event to the phone and then the phone answers. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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macOS urgent note for 24x7 folks on Sierra +
by Mario G 14 Aug '17

14 Aug '17

If you have FreeSWITCH running 24x7 on Sierra (and eventually High Sierra), you must update the plist in LaunchAgents to the updated one in the wiki: https://freeswitch.org/confluence/display/FREESWITCH/macOS+24x7+Preparation… <https://freeswitch.org/confluence/display/FREESWITCH/macOS+24x7+Preparation…> I spent months on this with Apple and instead of going into the very long story the short of it is: There was a design change (bug in my opinion) in 10.12.0, 10.12.1, 10.12.2, 10.12.5 and 10.12.6 (notice .3 and .4 were OK). Using the default process type macOS would not give CPU to FreeSWITCH even if nothing else was running. I tested this with other apps and it was not a problem, I could only make this happen with FreeSWITCH. I eventually found that overriding the process type resolved the issue in Sierra and High Sierra. Mario G

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ClueCon Talk: Open Source WebRTC testing suite, slides and resources
by Chad Phillips 14 Aug '17

14 Aug '17

I’ve had a few people email me privately about the content of my talk, so I figured I’d post a resource link: http://tiny.cc/cc-2017-webrtc-testing That contains the slides from the presentation, and a resource section with links to relevant projects/demos. Chad

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