January 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
by John Covici 12 Jan '20

12 Jan '20

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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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

12 Jan '20

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> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)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> > To: freeswitch-users(a)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 > 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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portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
by Tamer Higazi 11 Jan '20

11 Jan '20

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

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WebRTC STUN request for a SIP profile is not working
by Alexander Haugg 09 Jan '20

09 Jan '20

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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uuid_simplify after intercept/bridge
by Mitchell Langs 07 Jan '20

07 Jan '20

Hi, I need to originate 2 calls and later bridge them. Then Freeswitch needs to be removed from the call path. uuid_simplify accomplishes this. But where do I put this command so that it is executed once the two calls are bridged? As far as I understand the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge" command until the bridge is terminated. I have also tried to set sip_auto_simplify=true but it had no effect. Here's what I am executing: originate {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620 testing XML public <extension name="testing"> <condition field="destination_number" expression="^testing$"> <action application="set" data="api_result=${bgapi originate {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/> ... <action application="playback" data="$${hold_music}"/> </condition> </extension> Kind Regards

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detect_speech unimrcp ... ends with 'Aborted (core dumped)'
by Dmitriy Borisov 05 Jan '20

05 Jan '20

Hi, All! I have a strange problem... I have a big Lua script to initiate a call. It starts detect_speech with > execute_on_media=detect_speech unimrcp ... Script initiate Leg B on leg A answer. And when leg B answers FreeSWITCH halted with this message > 2019-12-25 00:00:55.903734 [INFO] switch_ivr_async.c:219 Digit parser > DPTOOLS: Setting realm to 'temp' > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_async.c:344 Digit parser > DPTOOLS: binding *#/temp/0 callback: 0x7fb375cf9700 data: 0x7fb2fc119818 > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1672 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State Change > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) Running State Change > CS_CONSUME_MEDIA (Cur 2 Tot 2) > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1063 sofia/internal/ > 1-FROM_GEN@10.17.19.111:5060 CUSTOM HOLD > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State CONSUME_MEDIA going > to sleep > freeswitch: src/switch_buffer.c:348: switch_buffer_zero: Assertion > `buffer->data != ((void *)0)' failed. > Aborted (core dumped) > I need to fix it ASAP, but I don't know where to see at all! What is it can be? -- With best regards Dmitry Borisov

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calling sleep after answer results in no audio
by Babak Yakhchali 04 Jan '20

04 Jan '20

Hi trying below dialplan results in no audio: <extension name="DailyIVR" continue="false" uuid="d8037520-2a15-11ea-9a9f-9bf4580ab5b8"> <condition field="destination_number" expression="^110$"> <action application="answer" data=""/> <action application="sleep" data="1000"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ringback=local_stream://default"/> <action application="set" data="transfer_ringback=local_stream://default"/> <action application="set" data="ivr_menu_uuid=d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/> <action application="ivr" data="d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/> <action application="" data=""/> </condition> </extension> but if I move *sleep before answer* everything works fine. Using tcpdump shows that when sleep is called after answer no rtp is established but if I move sleep before answer rtp is established I'm using freeswitch 1.8.7 on debian 8 thanks

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Re: [Freeswitch-users] No audio for conference call
by David P 02 Jan '20

02 Jan '20

You need a more detailed log, which on linux is usually at /var/log/freeswitch/freeswitch.log IIRC. Search your file system for this filename. You might also need to increase your logging level; do a web search for: freeswitch log level

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bypass_media_after_bridge
by Oleg Belousov 01 Jan '20

01 Jan '20

Happy festive season. Would you please guide me with usage of 'bypass_media_after_bridge' feature? Call flow is as following (running in lua script): 1. freeswitch preanswer call from UAC, then play some prompts 2, create second_session 3. when second session is ready - assign bypass_media_after_bridge=true and, see var is assigned. EXECUTE [depth=0] sofia/external/XXXXXXXXXX set(bypass_media_after_bridge=true) 2019-12-31 14:21:19.253279 [DEBUG] mod_dptools.c:1672 SET sofia/external/XXXXXXXXXX [bypass_media_after_bridge]=[true] 4. bridge both calls. At that time expect freeswtch fire re-invite to get out of rtp path, however do not see any re-invites. extract from script: ---- ConnectStr = "sofia/gateway/aed8dd2f-202f-4093-9851-436e21bafcf4/"..DN second_session = freeswitch.Session(ConnectStr) if (second_session:ready()) then second_session:execute("set","bypass_media_after_bridge=true") freeswitch.bridge(session, second_session) --- What could be wrong in my script? version: FreeSWITCH Version 1.10.1-release-12 -- Regards, Oleg

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Re: [Freeswitch-users] No audio for conference call
by David P 01 Jan '20

01 Jan '20

Search your log for 'Choose rtp'; the IP number following this shows what FS will use for a media stream. There is a line like this for each stream. For your scenario, you want these not to be private IPs. If they are private, you can try to auto-reject them by changing acl.conf.xml. But you'll have limited success due to this bug https://github.com/signalwire/freeswitch/issues/157

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