September 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Compact headers
by David Villasmil 28 Sep '20

28 Sep '20

Hello guys, Does freeswitch support mixing of compact and long headers? Thanks -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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CTI Web Application for Freeswitch
by Malik Ehtasham 28 Sep '20

28 Sep '20

Hi We would like to develop the Freeswitch CTI Web Application, from where a user can control all the CTI Features for example -Answer Call -hold/resume call -Consult/Transfer/Consult Transfer/Conference -Outbound Call I am looking for a library such as Cisco CUCM provides JTAPI <https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/jtapi_dev/10_5_2/C…> Do we have something similar in the Freeswitch, For now We are exploring Mod_commands <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands> , can this be used to acheive such a functionality, Though we tried few of its commands to answer the 2 phones but it didn't work. if mod_commands is not the option than what other option or track we should follow to acheive this . Any help/hint would be really helpful !! Thank you -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Distorted sound for one of conference members when AMR-WB codec used
by Rafał Telaczyński 28 Sep '20

28 Sep '20

Hi All, We are using FreeSWITCH to put 2 members on a conference and record it. The problem is that when AMR-WB codec is used, the second member to join the conference hears distorted sound(sometimes it lasts couple seconds, in some cases it lasts until the end of the call). There is no such sound heard by first member participant, nor is it generated by it(verified by decoding audio from RTP packets sent from first member participant towards FreeSWITCH). Moreover, distorted sound to second member can be heard after decoding RTP packets sent from FreeSWITCH towards second conference participant - proof that it is not the issue with second participant phone or so. Intrestingly, there is no such issue when AMR codec is used(tested by removing AMR-WB from global_codec_prefs) - the sound is clear for both participant throughout the whole duration of the conference. Considering that everything is OK with AMR codec used, it seems like the issue is not with conference itself, only using AMR-WB during conference. We tested with some of the FreeSWITCH parameters related to RTP(e.g. auto-rtp-bugs, rtp-rewrite-timestamps, send_silence_when_idle, etc.) but it did not help. Is it known issue? Did anyone faced something similar before? What else can be done to make it work? Please, let me know if you have any questions regarding the topic/need more info from my side. Thanks, Rafal

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Call Center
by Nathan Stratton 28 Sep '20

28 Sep '20

Any way to log off an agent if we lose SIP registration? ><> nathan stratton

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FS Not restarting RTP after unhold
by David Villasmil 25 Sep '20

25 Sep '20

Hello all, I have a scenario where FS is not restaring RTPs after an UNHOLD reInvite. The scenario is like this: FS--------------------------------------Endpoint <<<<<<<<<<<<< established call >>>>>>>>>>>>>>> |<---------- INVITE (SDP sendonly) ----------| |------------------ Trying -------------->| |----------- 200 OK (SDP recvonly)---------->| |<------------------ ACK -----------------| <<<<<<<<<<<<<<<<< ALL GOOD >>>>>>>>>>>>>>>>>>> |<------------ INVITE (no SDP) --------------| |------------------ Trying -------------->| |----------- 200 OK (SDP sendrcv) ---------->| |<----------- ACK (SDP sendrcv) -----------| But freeSWITCH never starts sending RTP after the UNHOLD... Don't know why. Anyone encountered this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Hide outbound trunk prefix from A-Leg
by Mateusz Bartczak 25 Sep '20

25 Sep '20

Hi I have simple scenario with FS bridging internal user with external SIP provider With SIP provider we agreed on tech prefix to distinguish between "service levels" Let's say trunk prefix is 9999 and user dials number 500500500 The problem is that when a local user is calling, he dials 500500500, the call is bridged to SIP provider with prepended tech prefix. After the call is established user will see on his softphone that the call is established with number 9999500500500 This creates some confusion, since they dialed number without prefix How can we hide the outbound call prefix and show the A-leg user only the number he dialed?

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Set a domain name in Request-URI without DNS queries
by Igor Potjevlesh 25 Sep '20

25 Sep '20

Hello, I try to setup a config with a domain name in the request-uri but I don't want Freeswitch to do a DNS request on it. I would like to send the INVITE with the distributor module to two gateways with their own IP in the realm param. Since now, every setup I tried with a realm or/and sip_invite_req_uri failed. Any advice? Thank you. Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

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RFC 4579 question
by Ryan Delgrosso 25 Sep '20

25 Sep '20

Hi All, Im working on getting RFC4579 conference support working, and when we create the room, i see the contact come back with isfocus on the 200ok (this indicates to me, along with the console output that trhe RFC-4579 flag is set properly on the room, unless im misreading something), but the subscription never generates a notify with the room participants (subscribe/200ok below for reference). Ive tried creating the room both with or without the @domain and neither makes a difference. The subscribe never gets a notify. Can someone point me in the right direction? Version is 1.10.2 on CentOS SUBSCRIBE sip:conf-c7250a4c-edf5-427f-af41-f7f61c251253@test.domain.com SIP/2.0 Via: SIP/2.0/UDP 172.31.0.110:6060;branch=z9hG4bKec61.cf48fb31.0;i=872400d6 To: <sip:conf-c7250a4c-edf5-427f-af41-f7f61c251253@test.domain.com> From: "rdelgrosso" <sip:rdelgrosso@test.domain.com>;tag=k5c2tba943 CSeq: 1 SUBSCRIBE Call-ID: 8806c2b0sphfb5rvq8s6 Max-Forwards: 69 Accept: application/conference-info+xml Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Event: conference Expires: 3600 Contact: <sip:172.31.0.110:6060;thinfo=VG98bzAdIFswMjIINB42BhBWZX1lWWhBdlpmUGlnYVhkXngAIgA9ICQAIht+AyMSRVMjHCNVckNiT2Biel9+XnJEalBjZ2Bc> Supported: outbound, replaces User-Agent: RingRx 0.1.0 Content-Length: 0 SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.31.0.110:6060;branch=z9hG4bKec61.cf48fb31.0;i=872400d6;rport=6060 From: "rdelgrosso" <sip:rdelgrosso@test.domain.com>;tag=k5c2tba943 To: <sip:conf-c7250a4c-edf5-427f-af41-f7f61c251253@test.domain.com>;tag=DsDlkFXvDBOl Call-ID: 8806c2b0sphfb5rvq8s6 CSeq: 1 SUBSCRIBE Contact: <sip:conf-c7250a4c-edf5-427f-af41-f7f61c251253@172.31.0.130:5080> Expires: 3600 User-Agent: RingRX Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: outbound, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0

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Re: [Freeswitch-users] Playback does not execute until first RTP
by David P 24 Sep '20

24 Sep '20

This is very interesting, because it's the reverse of a problem I've been asking about. Is there a way to trigger verto to send packets immediately after the DTLS handshake finishes, so FS can use them for "peer-reflexive" routing back? Clientside pcap's show that in some networks, verto/Chrome/network doesn't send anything to FS once the handshake is completed. ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 22 Sep 2020 10:25:34 -0500 > Subject: Re: [Freeswitch-users] Playback does not execute until first RTP > It's more than that, this is a standoff, each end is waiting for the > other, so if you use execute_on_answer01=playback::silence_stream://500 it > should get things moving. > > /b > > > On Tue, Sep 22, 2020 at 8:29 AM Jurijs Ivolga <jurijs.ivolga(a)gmail.com> > wrote: > >> Hi, >> >> This is strange, I remember some time ago I tested this with one of my >> buggy provider, which wasn't sending RTP until it received anything and >> after setting "disable-rtp-auto-adjust=true" in the SIP profile Freeswitch >> started to send RTP first and then SIP provider replied with RTP too. >> >> Have you tried to put this in SIP profile configuration? >> >> This was quite some time ago, so maybe I'm mixing it up. >> >> Jurijs >> >> >> On Tue, Sep 22, 2020 at 2:05 PM Oleg Belousov <obelousov(a)gmail.com> >> wrote: >> >>> Hi. >>> Thank you. >>> Yes, I tried to set up that parameter in the profile as true, but that >>> did not help. As far as I understand that is applicable for nated UAC, in >>> my scenario there is no nat. >>> -- >>> obelousov.tel >>> >>> >>> On Tue, Sep 22, 2020 at 12:01 PM Jurijs Ivolga <jurijs.ivolga(a)gmail.com> >>> wrote: >>> >>>> Hi, >>>> >>>> Have you checked this: >>>> >>>> >>>> https://lists.freeswitch.org/pipermail/freeswitch-users/2018-April/129500.h… >>>> >>>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/disable_rtp_auto_adjust >>>> >>>> Please try: >>>> >>>> disable-rtp-auto-adjust=false >>>> >>>> With kind regards, >>>> >>>> Jurijs >>>> >>>> >>>> On Tue, Sep 22, 2020 at 11:43 AM Oleg Belousov <obelousov(a)gmail.com> >>>> wrote: >>>> >>>>> Hi. >>>>> Met issue with prompt play (in lua, preanswer mode) - for certain >>>>> reasons freeswitch wait for RTP packet from UAC side, and only after that >>>>> started to play prompt. If UAC does not send RTP, the script does not >>>>> execute further. Is there a way to instruct Freeswitch to send RTP packets >>>>> first by itself? >>>>> -- >>>>> obelousov.tel >>>>> >>>>>

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Block DTMF audio tones (non-RFC2833) on bridged calls?
by Andy Newlands 24 Sep '20

24 Sep '20

I want to prevent DTMF audio tones (I.E. not RFC2833 or INFO) from being passed across to the other leg of a bridged call, while still being able to detect them at an application level. I have no problem detecting the digits - "start_dtmf" works just fine for that. The problem is stopping the tones from being heard on the opposite call leg. I have written a simple FS application module to capture and process tones, and this works well in all scenarios for my application with RFC2833 DTMF (fyi: I use switch_core_event_hook_add_recv_dtmf() ), but not when the incoming call is sending audio DTMF tones - the app detects them fine, but the other party also hears them. I can't see any way of achieving this using standard DP apps/vars. I've also looked (high level) through the source code around media bugs and in switch_ivr_bridge and can't see any obvious place to stop the DTMF tone frames from getting through the bridge. In fact, it seems to me (at this point in time) that maybe the frames are sent before DTMF detection takes place (is that true?) - I really wish there was more documentation around the source code :-) If anyone can tell me how to achieve this using standard DP apps/vars, that would be great. However, I'm quite happy to make code changes - in my module or the FS code - to get this working, if someone could just point me in the right direction... NOTE: We have no control over the use of audio tones for DTMF - this is forced upon us by an external provider (TalkTalk - UK). Kind regards, Andy

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