- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Pause/resume recording mod_callcenter
by Jonathan Hunter 24 Sep '20

24 Sep '20

Hi guys, I did see a post some years back when someone was using mod_callcenter and trying to get pause/resume recording working correctly. I am aware callcenter records by default but have removed that from the profile, and I am trying to set the record parameters below before it hits the call queue; table.insert(xml, [[<action application="export" data="RECORD_STEREO=true"/>]]); table.insert(xml, [[<action application="export" data="RECORD_TOGGLE_ON_REPEAT=true"/>]]); table.insert(xml, [[<action application="export" data="RECORD_APPEND=true"/>]]); table.insert(xml, [[<action application="export" data="execute_on_answer=record_session $${recordings_dir}/${caller_id_number}.${uuid}.wav"/>]]); table.insert(xml, [[<action application="bind_meta_app" data="1 b s stop_record_session::$${recordings_dir}/${caller_id_number}.${uuid}.wav"/>]]); table.insert(xml, [[<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${uuid}.wav"/>]]); When the call answers the recording is there, however when I hit *1 it does stop the recording, however when I hit *2 it starts recording again but creates a new file, doesnt append the existing file. Am I doing something wrong here with configuration? As just want to be able to pause recording, and then start it which would then append. We are using freeSWITCH version 1.10.5. Any advice would be great! Thanks Jon

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Is it good idea to split FS core and Sofia read and write queries
by sagar malam 23 Sep '20

23 Sep '20

Hello All , I am wondering if it is a good idea to split FS core and mod_sofia read and write queries among master(write only) and slave(read only) via ProxySQL ? JFI, I am using MariaDB Galera Cluster. -- Thanks, Sagar

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MID in SDP
by David Villasmil 23 Sep '20

23 Sep '20

Hello all, I have this scenario where a cisco puts the call on hold, then gets it out of hold. (using recvonly and sendrecv). Then this cisco sends a reInvite without SDP and FS replies with 200 OK which includes that "mid: audio" but never sends RTP... anyone ever encounter this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Playback does not execute until first RTP
by Oleg Belousov 23 Sep '20

23 Sep '20

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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Verto Settings
by Juan E 22 Sep '20

22 Sep '20

I would like to use the benefits of Verto WebRTC with my IP central. I would like to register my extensions in the same way as in the demo, but in my central IP. Is that possible?? -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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