January 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Announcing conference participant after PIN entry
by Thilo-Alexander Ginkel 27 Jan '20

27 Jan '20

Hello everyone, let's say, I have a FreeSWITCH conference that is PIN-protected and where I want to record each participant's name. How do I announce the name of a participant *after* the caller's PIN has been validated? So far I tried something like this: <extension name="Record Name and schedule conf announce"> <condition field="destination_number" expression="^34(\d{2})$"> <action application="answer"/> <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-record_name1.wav"/> <action application="playback" data="tone_stream://%(500,0,500)"/> <action application="record" data="${namefile} 5"/> <action application="playback" data="ivr/ivr-call_being_transferred.wav"/> <action application="set" data="res=${sched_api +1 none conference 31$1-${domain_name} play file_string://${namefile}!conference/conf-has_joined.wav}"/> <action application="transfer" data="31$1 XML default"/> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband${conference_pin}"/> </condition> </extension> Participant dials 34xx, name is recorded, playback scheduled, call transferred to 31xx, which initiates the conference, but sched_api plays the join message before the pin has been validated. Is there a way to hook into the conference so that the join message is only scheduled after the PIN has been validated successfully? Thanks, Thilo

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Grandstream ATA702 - dropped calls
by William Colls 25 Jan '20

25 Jan '20

When I make an outgoing call through this device, the call is often disconnected after about 30-40 seconds. Incoming calls are fine. I have SIP transport set to UDP, NAT Transversal set to NO, Enable SIP Options Keel Alive set to YES, SIP Options Keep Alive Interval set at 30, and SIP Options Keep Alive Max Lost at 3. Freeswitch version 1.10.1 was started with -nonat. Obviously I have missed, or miss-understood, something. Any pointers or suggestions will be most welcome. Thanks for your time. William.

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Cannot force transcoding between PCMU and PCMA
by Kliment Toshkov, Netfinity JSC 23 Jan '20

23 Jan '20

Hello, I have to force transcoding between PCMU and PCMA in the following scenario: Customer [PCMA, PCMU, G729] > Kamailio > Freeswitch 1..N > termination gateway [PCMA] I need transcoding from PCMU and G729 to take place on Freeswitch boxes. Currently it works for G729 but for PCMU calls they are originated with PCMU. Inbound profile: <param name="inbound-codec-prefs" value="$${global_codec_prefs}”/> <param name="disable-transcoding" value="false"/> Outbound profile: <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="disable-transcoding" value="false"/> vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/> In xml dial plan: <action application="set" data="inherit_codec=false"/> <action application="set" data="media_mix_inbound_outbound_codecs=true"/> But the calls are still not transcoded between PCMU and PCMA. Only one thing worked - forcing the codec in bridge app: <action application="bridge" data="{absolute_codec_string='PCMA'} ..."/> But I really need to know what I am missing and how to fix it using config files only. Brute force is not preferred. Thank you! Kliment Toshkov Key Accounts 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

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No audio for conference call
by Md,Mehedi Hasan Kabir(Tanim) 23 Jan '20

23 Jan '20

Hi I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this? Audio is fine when I call other users directly. Regards Tanim

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FreeSWITCH uniMRCP dynamic speech context
by Arjunstl 23 Jan '20

23 Jan '20

Hi, I'm working on ASR application with uniMRCP in lua. but when doing with dynamic speech context mentioned in unimrcp usage manual. i'm unable to get exact results. my aim is to recognize only those words which i want to define in phrases(my defined speech context). most of the times, recognizer recognizes random words, that are not defined in speech context. any help or suggestion would be highly appreciated. Best Regards -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Zoiper and push notification
by Alex Lake 22 Jan '20

22 Jan '20

Hi, Is anyone here using Zoiper mobile app and their push notification system? I had a go at following their instructions, but I don't think it worked. Cheers Alex

2 1

How to generate only one CDR record regardless of transfer?
by Chhorm Chhatra 21 Jan '20

21 Jan '20

Hi FreeSWITCH mailing list, Currently, I encountered some problem when the call get transferred, freeswitch generated multiple CDR for it. How could I configure freeswitch to only create one CDR for both leg A and B even if it get transferred? Any help would be really appreciated. Thank you in advance. best regards, Chhorm Chhatra

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Re: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working
by Alexander Haugg 14 Jan '20

14 Jan '20

Hi, if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and use this variable $${external_rtp_ip} in the SIP Profile, then the Ext-RTP-IP show the real external IP. I think, that’s OK. But I miss the srflx candidate in the sdp. I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works. What’s wrong? Here is my sofia profile config: <profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua"> <aliases /> <domains> <domain name="h3k.sip.c4b" alias="false" parse="true" /> </domains> <settings> <param name="user-agent-string" value="XPhone Call Controller" /> <param name="debug" value="0" /> <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="sip-trace" value="no" /> <param name="sip-capture" value="no" /> <param name="dialplan" value="XML" /> <param name="context" value="SBC2_SP_outbound" /> <param name="dtmf-type" value="rfc2833" /> <param name="dtmf-duration" value="100" /> <param name="rfc2833-pt" value="98" /> <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" /> <param name="hold-music" value="local_stream://moh" /> <param name="rtp-timer-name" value="none" /> <param name="rtp-rewrite-timestamps" value="true" /> <param name="manage-presence" value="false" /> <param name="inbound-codec-negotiation" value="generous" /> <param name="inbound-late-negotiation" value="true" /> <param name="nonce-ttl" value="60" /> <param name="auth-calls" value="false" /> <param name="sip-port" value="4901" /> <param name="rtp-ip" value="MyLocalIP" /> <param name="sip-ip" value=" MyLocalIP " /> <param name="ext-rtp-ip" value="$${external_rtp_ip}" /> <param name="ext-sip-ip" value=" MyLocalIP " /> <param name="local-network-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="rfc1918.auto" /> <param name="apply-candidate-acl" value="SBC2_localnet.auto" /> <param name="apply-candidate-acl" value="any_v4.auto" /> <param name="apply-candidate-acl" value="wan_v4.auto" /> <param name="apply-inbound-acl" value="SBC2_localnet.auto" /> <param name="rtp-timeout-sec" value="300" /> <param name="rtp-hold-timeout-sec" value="1800" /> <param name="enable-3pcc" value="true" /> <param name="inbound-use-callid-as-uuid" value="true" /> <param name="tls" value="false" /> <param name="tls-only" value="false" /> <param name="tls-bind-params" value="transport=tls" /> <param name="tls-sip-port" value="" /> <param name="tls-passphrase" value="" /> <param name="tls-verify-date" value="true" /> <param name="tls-verify-depth" value="2" /> <param name="tls-verify-in-subjects" value="" /> <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" /> <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" /> <param name="zrtp-passthru" value="true" /> <param name="inbound-reg-force-matching-username" value="true" /> <param name="auth-all-packets" value="false" /> <param name="wss-binding" value=":4447" /> <param name="caller-id-type" value="rpid" /> <param name="manual-redirect" value="true" /> <param name="rtcp-audio-interval-msec" value="5000" /> </settings> <gateways> <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" /> </gateways> </profile> Von: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> Im Auftrag von Alexander Haugg Gesendet: Donnerstag, 9. Januar 2020 08:17 An: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Betreff: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working Hi, I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting. <param name="stun-enabled" value="true"/> <param name="stun-auto-disable" value="false"/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org" /> The Freeswitch is started with the parameter –nonat In the pcap trace is no STUN request visible. I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request. If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully. The status output of the profile is: ================================================================================================= Name SBC2_B2Bua Domain Name N/A Auto-NAT false DBName sofia_reg_SBC2_B2Bua Pres Hosts Dialplan XML Context SBC2_SP_outbound Challenge Realm auto_to RTP-IP MY_LOCAL_IP Ext-RTP-IP stun:stun.freeswitch.org SIP-IP MY_LOCAL_IP Ext-SIP-IP MY_LOCAL_IP URL sip:mod_sofia@MY_LOCAL_IP:4901 BIND-URL sip:mod_sofia@MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,PCMU,PCMA,VP8 CODECS OUT OPUS,PCMU,PCMA,VP8 TEL-EVENT 98 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 Thanks a lot!!! Alex

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Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION (SOLVED)
by David Villasmil 12 Jan '20

12 Jan '20

On the cli: sofia profile [internal/external] siptrace on Where internal/external is the profile you’re using. Should get the trace on the cli itself. On Sun, 12 Jan 2020 at 15:02, Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 12 Jan 2020 15:28:01 +0100 > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION (SOLVED) > Yes....of course! > > I try to simulate it once again with PulseAudio in combination with > mod_portaudio. > > In the meantime, could you please tell me how to raise the logginglevel > and debugging for you and where it is saved? > > I think logging on my machine is through mod_cdrcsv log. > > > Thanks. > > On 2020-01-12 15:16, David Villasmil wrote: > > Can you provide more logging? And also a trace? > > > > On Sun, 12 Jan 2020 at 14:00, Tamer Higazi via FreeSWITCH-users > > <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> wrote: > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi <th982a(a)googlemail.com > > <mailto:th982a@googlemail.com>> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > Cc: > > Bcc: > > Date: Sun, 12 Jan 2020 14:49:46 +0100 > > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on > > FS, but INCOPATIBLE DESTINATION (SOLVED) > > Dear John, > > > > Thank you very much for your advise. > > It works very well! > > > > It works only good and without any problems, if I choose the device > > directly. > > If I choose "pulse" as ringdev, outdev and indev I have this > > "cracking" > > perioudicly in the line. > > > > > > Thanks, Tamer > > > > On 2020-01-12 08:06, John Covici wrote: > > > How about <action application="export" > > > data="media_mix_inbound_outbound_codecs=true"/> See if that works. > > > > > > On Sat, 11 Jan 2020 19:52:54 -0500, > > > Tamer Higazi via FreeSWITCH-users wrote: > > >> [1 <message/rfc822 (7bit)>] > > >> To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > >> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works > > on FS, but INCOPATIBLE DESTINATION > > >> From: Tamer Higazi <th982a(a)googlemail.com > > <mailto:th982a@googlemail.com>> > > >> Date: Sun, 12 Jan 2020 01:16:57 +0100 > > >> > > >> Dear David, > > >> > > >> This is what I thought as well (just shortly). > > >> > > >> with the SIP phone dialing it bridges through the gateway, it > > >> works without any problems. > > >> > > >> But with mod_portaudio or mod_alsa I have this problem. > > >> > > >> Any ideas how to fix it or to get more informations or how to > > fix it ? > > >> > > >> Because in the FS pages it's written nowhere what codecs should > > >> be used or how to configure it .... > > >> > > >> > > >> best, Tamer > > >> > > >> On 2020-01-12 01:12, David Villasmil wrote: > > >>> That usually comes up when the far side and fs can’t agree on a > > >>> codec. Make sure both sides have the same set of codecs > > >>> enabled. > > >>> > > >>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users > > >>> <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > >>> <mailto:freeswitch-users@lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>>> wrote: > > >>> > > >>> > > >>> > > >>> > > >>> ---------- Forwarded message ---------- > > >>> From: Tamer Higazi <th982a(a)googlemail.com > > <mailto:th982a@googlemail.com> > > >>> <mailto:th982a@googlemail.com > > <mailto:th982a@googlemail.com>>> > > >>> To: freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > >>> <mailto:freeswitch-users@lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > >>> Cc: > > >>> Bcc: > > >>> Date: Sat, 11 Jan 2020 22:21:56 +0100 > > >>> Subject: portaudio and pulsaudio works on FS, but > INCOPATIBLE > > >>> DESTINATION > > >>> Hi people, > > >>> > > >>> After long try and there I got "finally" portaudio > > running on my cpu. > > >>> > > >>> I got the dumb error message "connection refused" rid of > > me, and FS > > >>> seems to access the soundcard. > > >>> I can choose the devices of my choice, and place a call. > > >>> > > >>> But when I place a call, I still get this: > > >>> > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_core_state_machine.c:935 > > >>> (sofia/external/123456) State REPORTING > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_core_state_machine.c:174 > > >>> sofia/external/123456 Standard REPORTING, cause: > > >>> INCOMPATIBLE_DESTINATION > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_core_state_machine.c:935 > > >>> (sofia/external/123456) State REPORTING going to sleep > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_core_state_machine.c:610 > > >>> (sofia/external/123456) State Change CS_REPORTING -> > > CS_DESTROY > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > switch_core_session.c:1715 > > >>> Session 17 > > >>> (sofia/external/123456) Locked, Waiting on external entities > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_ivr_originate.c:3941 > > >>> Originate > > >>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > > >>> 2020-01-11 22:18:33.847800 [NOTICE] > > switch_core_session.c:1733 > > >>> Session > > >>> 17 (sofia/external/123456) Ended > > >>> 2020-01-11 22:18:33.847800 [NOTICE] > > switch_core_session.c:1737 Close > > >>> Channel sofia/external/123456 [CS_DESTROY] > > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047 > > >>> (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE > > >>> 2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 > > Originate > > >>> Failed. > > >>> Cause: INCOMPATIBLE_DESTINATION > > >>> 2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 > > Hangup > > >>> portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > >>> 2020-01-11 22:18:33.847800 [DEBUG] > > switch_core_state_machine.c:738 > > >>> (sofia/external/123456) Running State Change CS_DESTROY > > (Cur 1 Tot 17) > > >>> > > >>> Any ideas why this is the case ? > > >>> > > >>> > > >>> best, Tamer > > >>> > > >>> > > >>> > > >>> > > >>> > > >>> ---------- Forwarded message ---------- > > >>> From: Tamer Higazi via FreeSWITCH-users > > >>> <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > >>> <mailto:freeswitch-users@lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>>> > > >>> To: freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org> > > >>> <mailto:freeswitch-users@lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > >>> Cc: > > >>> Bcc: > > >>> Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST) > > >>> Subject: [Freeswitch-users] portaudio and pulsaudio works > > on FS, > > >>> but INCOPATIBLE DESTINATION > > >>> > > > _________________________________________________________________________ > > >>> > > >>> The FreeSWITCH project is sponsored by SignalWire > > >>> https://signalwire.com > > >>> Enhance your FreeSWITCH install with disruptive priced > > SMS and > > >>> PSTN services. > > >>> Build your next product on our scalable cloud platform. > > >>> > > >>> Join our online community to chat in real time > > >>> https://signalwire.community > > >>> > > >>> Professional FreeSWITCH Services > > >>> sales(a)freeswitch.com <mailto:sales@freeswitch.com> > > <mailto:sales@freeswitch.com <mailto:sales@freeswitch.com>> > > >>> https://freeswitch.com > > >>> > > >>> Official FreeSWITCH Sites > > >>> https://freeswitch.com/oss > > >>> https://freeswitch.org/confluence > > >>> https://cluecon.com > > >>> > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users(a)lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > >>> <mailto:FreeSWITCH-users@lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> https://freeswitch.com > > >>> > > >>> -- > > >>> Regards, > > >>> > > >>> David Villasmil > > >>> email: david.villasmil.work(a)gmail.com > > <mailto:david.villasmil.work@gmail.com> > > >>> <mailto:david.villasmil.work@gmail.com > > <mailto:david.villasmil.work@gmail.com>> > > >>> phone: +34669448337 > > >> [2 <text/plain; utf-8 (base64)>] > > >> > > > _________________________________________________________________________ > > >> > > >> The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > >> Enhance your FreeSWITCH install with disruptive priced SMS and > > PSTN services. > > >> Build your next product on our scalable cloud platform. > > >> > > >> Join our online community to chat in real time > > https://signalwire.community > > >> > > >> Professional FreeSWITCH Services > > >> sales(a)freeswitch.com <mailto:sales@freeswitch.com> > > >> https://freeswitch.com > > >> > > >> Official FreeSWITCH Sites > > >> https://freeswitch.com/oss > > >> https://freeswitch.org/confluence > > >> https://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 > > >> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> https://freeswitch.com > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi via FreeSWITCH-users > > <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org > > <mailto:freeswitch-users@lists.freeswitch.org>> > > Cc: > > Bcc: > > Date: Sun, 12 Jan 2020 06:00:54 -0800 (PST) > > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on > > FS, but INCOPATIBLE DESTINATION (SOLVED) > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and > > PSTN services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://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 > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work(a)gmail.com > > <mailto:david.villasmil.work@gmail.com> > > phone: +34669448337 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 12 Jan 2020 07:02:41 -0800 (PST) > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION (SOLVED) > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION (SOLVED)
by David Villasmil 12 Jan '20

12 Jan '20

Can you provide more logging? And also a trace? On Sun, 12 Jan 2020 at 14:00, Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 12 Jan 2020 14:49:46 +0100 > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION (SOLVED) > Dear John, > > Thank you very much for your advise. > It works very well! > > It works only good and without any problems, if I choose the device > directly. > If I choose "pulse" as ringdev, outdev and indev I have this "cracking" > perioudicly in the line. > > > Thanks, Tamer > > On 2020-01-12 08:06, John Covici wrote: > > How about <action application="export" > > data="media_mix_inbound_outbound_codecs=true"/> See if that works. > > > > On Sat, 11 Jan 2020 19:52:54 -0500, > > Tamer Higazi via FreeSWITCH-users wrote: > >> [1 <message/rfc822 (7bit)>] > >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > >> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, > but INCOPATIBLE DESTINATION > >> From: Tamer Higazi <th982a(a)googlemail.com> > >> Date: Sun, 12 Jan 2020 01:16:57 +0100 > >> > >> Dear David, > >> > >> This is what I thought as well (just shortly). > >> > >> with the SIP phone dialing it bridges through the gateway, it > >> works without any problems. > >> > >> But with mod_portaudio or mod_alsa I have this problem. > >> > >> Any ideas how to fix it or to get more informations or how to fix it ? > >> > >> Because in the FS pages it's written nowhere what codecs should > >> be used or how to configure it .... > >> > >> > >> best, Tamer > >> > >> On 2020-01-12 01:12, David Villasmil wrote: > >>> That usually comes up when the far side and fs can’t agree on a > >>> codec. Make sure both sides have the same set of codecs > >>> enabled. > >>> > >>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users > >>> <freeswitch-users(a)lists.freeswitch.org > >>> <mailto:freeswitch-users@lists.freeswitch.org>> wrote: > >>> > >>> > >>> > >>> > >>> ---------- Forwarded message ---------- > >>> From: Tamer Higazi <th982a(a)googlemail.com > >>> <mailto:th982a@googlemail.com>> > >>> To: freeswitch-users(a)lists.freeswitch.org > >>> <mailto:freeswitch-users@lists.freeswitch.org> > >>> Cc: > >>> Bcc: > >>> Date: Sat, 11 Jan 2020 22:21:56 +0100 > >>> Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE > >>> DESTINATION > >>> Hi people, > >>> > >>> After long try and there I got "finally" portaudio running on my > cpu. > >>> > >>> I got the dumb error message "connection refused" rid of me, and > FS > >>> seems to access the soundcard. > >>> I can choose the devices of my choice, and place a call. > >>> > >>> But when I place a call, I still get this: > >>> > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935 > >>> (sofia/external/123456) State REPORTING > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174 > >>> sofia/external/123456 Standard REPORTING, cause: > >>> INCOMPATIBLE_DESTINATION > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935 > >>> (sofia/external/123456) State REPORTING going to sleep > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610 > >>> (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715 > >>> Session 17 > >>> (sofia/external/123456) Locked, Waiting on external entities > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941 > >>> Originate > >>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > >>> 2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733 > >>> Session > >>> 17 (sofia/external/123456) Ended > >>> 2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 > Close > >>> Channel sofia/external/123456 [CS_DESTROY] > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047 > >>> (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE > >>> 2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate > >>> Failed. > >>> Cause: INCOMPATIBLE_DESTINATION > >>> 2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup > >>> portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > >>> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738 > >>> (sofia/external/123456) Running State Change CS_DESTROY (Cur 1 > Tot 17) > >>> > >>> Any ideas why this is the case ? > >>> > >>> > >>> best, Tamer > >>> > >>> > >>> > >>> > >>> > >>> ---------- Forwarded message ---------- > >>> From: Tamer Higazi via FreeSWITCH-users > >>> <freeswitch-users(a)lists.freeswitch.org > >>> <mailto:freeswitch-users@lists.freeswitch.org>> > >>> To: freeswitch-users(a)lists.freeswitch.org > >>> <mailto:freeswitch-users@lists.freeswitch.org> > >>> Cc: > >>> Bcc: > >>> Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST) > >>> Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, > >>> but INCOPATIBLE DESTINATION > >>> > _________________________________________________________________________ > >>> > >>> The FreeSWITCH project is sponsored by SignalWire > >>> https://signalwire.com > >>> Enhance your FreeSWITCH install with disruptive priced SMS and > >>> PSTN services. > >>> Build your next product on our scalable cloud platform. > >>> > >>> Join our online community to chat in real time > >>> https://signalwire.community > >>> > >>> Professional FreeSWITCH Services > >>> sales(a)freeswitch.com <mailto:sales@freeswitch.com> > >>> https://freeswitch.com > >>> > >>> Official FreeSWITCH Sites > >>> https://freeswitch.com/oss > >>> https://freeswitch.org/confluence > >>> https://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 > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> https://freeswitch.com > >>> > >>> -- > >>> Regards, > >>> > >>> David Villasmil > >>> email: david.villasmil.work(a)gmail.com > >>> <mailto:david.villasmil.work@gmail.com> > >>> phone: +34669448337 > >> [2 <text/plain; utf-8 (base64)>] > >> > _________________________________________________________________________ > >> > >> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > >> Build your next product on our scalable cloud platform. > >> > >> Join our online community to chat in real time > https://signalwire.community > >> > >> Professional FreeSWITCH Services > >> sales(a)freeswitch.com > >> https://freeswitch.com > >> > >> Official FreeSWITCH Sites > >> https://freeswitch.com/oss > >> https://freeswitch.org/confluence > >> https://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 > >> https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 12 Jan 2020 06:00:54 -0800 (PST) > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION (SOLVED) > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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