- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Run script that calls exts and plays recording
by Vallimamod Abdullah 01 Jun '19

01 Jun '19

Hi, There is no b-leg when the call is answered by freeswitch. You need to create new sessions for your subsequent calls in your lua script. Best Regards, -- Vallimamod Abdullah SIP Solutions vma(a)sip.solutions linkedin.com/in/vallimamod . > On 1 Jun 2019, at 13:54, Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: Bipin Patel <bipin(a)xbipin.com <mailto:bipin@xbipin.com>> > Subject: Re: [Freeswitch-users] Run script that calls exts and plays recording > Date: 1 June 2019 at 12:34:19 CEST > To: <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I have a requirement where one phone calls the number 55 which fs answers and asks user to record his message and hangup when complete which works fine but after he hangs up I want to run a lua script which can take this recorded message and call multiple phones and play it to them but this part doesn't work even when using api_hangup_hook because fs replies not session available. > > Is there any way to keep the b leg which is fs itself answering active and run the script? It's fs answering and recording the message so not able to use other commands related to bridge like continue on fail or hang up after bridge or transfer etc. > > Any pointers will be helpful > > > > Regards, > > Bipin > > On June 1, 2019 1:51:09 PM Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> wrote: > >> _________________________________________________________________________ >> >> 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 > > > > > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/> > > Professional FreeSWITCH Services > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > https://freeswitch.com <https://freeswitch.com/> > > Official FreeSWITCH Sites > https://freeswitch.com/oss <https://freeswitch.com/oss> > https://freeswitch.org/confluence <https://freeswitch.org/confluence> > https://cluecon.com <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 <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> > https://freeswitch.com <https://freeswitch.com/>

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Re: [Freeswitch-users] Run script that calls exts and plays recording
by Bipin Patel 01 Jun '19

01 Jun '19

Hi, I have a requirement where one phone calls the number 55 which fs answers and asks user to record his message and hangup when complete which works fine but after he hangs up I want to run a lua script which can take this recorded message and call multiple phones and play it to them but this part doesn't work even when using api_hangup_hook because fs replies not session available. Is there any way to keep the b leg which is fs itself answering active and run the script? It's fs answering and recording the message so not able to use other commands related to bridge like continue on fail or hang up after bridge or transfer etc. Any pointers will be helpful Regards, Bipin On June 1, 2019 1:51:09 PM Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > _________________________________________________________________________ > > 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

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Run script that calls exts and plays recording
by Bipin Patel 01 Jun '19

01 Jun '19

Hi, I have a requirement where one phone calls the number 55 which fs answers and asks user to record his message and hangup when complete which works fine but after he hangs up I want to run a lua script which can take this recorded message and call multiple phones and play it to them but this part doesn't work even when using api_hangup_hook because fs replies not session available. Is there any way to keep the b leg which is fs itself answering active and run the script? It's fs answering and recording the message so not able to use other commands related to bridge like continue on fail or hang up after bridge or transfer etc. Any pointers will be helpful Regards, Bipin

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VoIP and WebRTC Trivia
by Abbi Minessale 31 May '19

31 May '19

Abbi Minessale <abbi(a)freeswitch.com> Thu, May 30, 3:26 PM (18 hours ago) to freeswitch-users Hey everyone, My name is Abbi Minessale and I am the content strategist for FreeSWITCH. This year at ClueCon we are adding a large-scale trivia game to our Hack-A-Thon and I need your help! In the Google form linked below please enter any trivia questions you can think of about VoIP, WebRTC, technology, FreeSWITCH, Open Source, ClueCon, programming in general, or any other techie current events! Thanks so much and I hope to see you at ClueCon this year! https://forms.gle/vuH2zACEh4Z6TkkZ8 -- Abbi Minessale | Content Strategist SignalWire: *228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303 <https://maps.google.com/?q=228+Hamilton+Ave+3rd+Floor,+Palo+Alto,+CA+94303&…>* Email: abbi(a)minessale.com Website: <https://www.freeswitch.com/>https://www.signalwire.com

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mod_com_amd
by Vishal Pai 31 May '19

31 May '19

Hello everyone I am using commercial module for AMD it works well with example given on the portal. Now my scenario is here in which i need some suggestion. Caller == Calls on DID == IVR == Call bridge to External Number (here we have amd check ) == If AMD detects then transfer to another number now if person is detected the calls should continue between Caller and External Number. Thanks

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creating conferences
by Amit Chaudhuri 31 May '19

31 May '19

Hi, I am trying to create a conference using the fs_cli in order to experiment with the conferencing capabilities. I can issue a 'conference list' command and get (as expected) a message confirming I have no active conferences. I can also issue 'conference xml_list', 'conference commands', 'conference help' and get sensible answers. When I try 'conference dial sofia/3001 12345 test' for example, I get a message: "Conference dial not found". Is it possible to create a conference from the cli? In particular, is it possible to create an empty conference that users can dial into at a later point in time? I've not found any examples which help me, although I have seen an unresolved bug report - FS11133 - that might imply one cannot create empty conferences. Any help appreciated. Regards, Amit

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sched_api survive restarts
by Bipin Patel 31 May '19

31 May '19

hi, is it possible to make sched_api tasks persistent and survive restarts, i want to create a wakeup call scenario where the admin sets a time and all extensions in a group are called at that time and played a sound file and admin can change time at will. I looked at a few lua scripts but mostly all make use of the sched_api and i noticed they dont survive a FS restart -- Regards, Bipin

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refusing t38 in outgoing fax server transmission
by Sandro Bordacchini 30 May '19

30 May '19

Hello everyone. I am have a few FS installations interconnected with a voip provider; during outbound fax server transmissions, they told us that we need *to refuse T38 even if offered by the network*. The sdp offer/answer negotiation varies depending on the called number, probably due to a different routing in the voip provider's network. I experience some dropped calls (even before the fax transmission kicks in) using both FS versions 1.4.9 and 1.6.16. This is a successful sdp negotiation (after call is estabilished) First reinvite: REMOTE INVITE with SDP: v=0..o=LucentPCSF 413445855 413445856 IN IP4 10.198.1.230..s=-..c=IN IP4 10.198.1.230..t=0 0..m=audio 15756 RTP/AVP 8 96 4..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-15..a=ptime:20..a=X-sqn: 0..a=X-cap: 1 image udptl t38.. LOCAL 200OK with SDP: v=0..o=FreeSWITCH 1558069426 1558069428 IN IP4 10.197.98.6..s=FreeSWITCH..c=IN IP4 10.197.98.6..t=0 0..m=audio 30394 RTP/AVP 8 96..a=rtpmap:8 PCMA/8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-16..a=silenceSupp:off - - - -..a=ptime:20.. Then, a second reinvite happens: REMOTE INVITE with SDP: v=0..o=LucentPCSF 413445855 413445857 IN IP4 10.198.1.230..s=-..c=IN IP4 10.198.1.230..t=0 0..m=audio 0 RTP/AVP 0..c=IN IP4 10.198.1.230..m=image 12834 udptl t38..a=X-sqn: 0..a=X-cap: 1 image udptl t38.. LOCAL 488 Then, the last reinvite: REMOTE INVITE with SDP: v=0..o=LucentPCSF 413445855 413445858 IN IP4 10.198.1.230..s=-..c=IN IP4 10.198.1.230..t=0 0..m=audio 15756 RTP/AVP 8 96 4..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-15..a=ptime:20..a=X-sqn: 0..a=X-cap: 1 image udptl t38..m=image 0 udptl t38..c=IN IP4 10.198.1.230.. LOCAL 200OK with SDP: v=0..o=FreeSWITCH 1558069426 1558069429 IN IP4 10.197.98.6..s=FreeSWITCH..c=IN IP4 10.197.98.6..t=0 0..m=audio 30394 RTP/AVP 8 96..a=rtpmap:8 PCMA/8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-16..a=silenceSupp:off - - - -..a=ptime:20..m=image 0 udptl t38.. And the fax is transmitted correctly This is an unsuccessful sdp negotiation: First reinvite REMOTE INVITE with SDP: v=0..o=LucentPCSF 579041281 579041282 IN IP4 10.198.1.230..s=-..c=IN IP4 10.198.1.230..t=0 0..m=audio 13302 RTP/AVP 8 96..a=rtpmap:96 telephone-event/8000..a=ptime:10..a=ecan:fb on -..a=Fax..m=image 17522 udptl t38..a=T38FaxVersion:0..a=T38MaxBitRate:14400..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:1800..a=T38FaxMaxDatagram:260..a=T38FaxUdpEC:t38UDPRedundancy..a=ecan:fb on -.. LOCAL 488 and the remote hangs up. My point is that, in case of unsuccessful negotiation, FS receives a reinvite with two "m=" in the sdp, one G711A (port 13302) and one T38 (port 17522). Is it ok for FS to refuse "all the sdp" with 488 just because it has to refuse T38 or it should be configured to refuse only the T38 media? Should remote endpoint send another reinvite with only G711 or only T38 before hanging up? Thanks in advance to whoever read till here :-) Any hints? S. P.s.: This is what i get in the log in the unsuccessful case during the reinvite (1.4.9 code): [...] 2019-05-17 11:22:57.498389 [DEBUG] switch_core_media.c:3476 Audio Codec Compare [PCMA:8:8000:10:64000:1]/[PCMA:8:8000:10:64000:1] 2019-05-17 11:22:57.498389 [DEBUG] switch_core_media.c:3531 Audio Codec Compare [PCMA:8:8000:10:64000:1] ++++ is saved as a match 2019-05-17 11:22:57.498389 [DEBUG] switch_core_media.c:3397 Set telephone-event payload to 96 2019-05-17 11:22:57.498389 [DEBUG] switch_core_media.c:3713 Set 2833 dtmf send payload to 96 2019-05-17 11:22:57.498389 [ERR] sofia.c:7065 Reinvite Codec Error! 2019-05-17 11:22:57.498389 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/<calledfaxnumber> [BREAK] 2019-05-17 11:22:57.498389 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/<calledfaxnumber> [BREAK] 2019-05-17 11:22:57.498389 [DEBUG] switch_core_media.c:2280 Already using PCMA 2019-05-17 11:22:57.498389 [DEBUG] sofia.c:6444 Channel sofia/external/<calledfaxnumber> entering state [ready][488] 2019-05-17 11:22:57.638443 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/<calledfaxnumber> [BREAK] 2019-05-17 11:22:57.638443 [NOTICE] sofia.c:952 Hangup sofia/external/<calledfaxnumber> [CS_EXECUTE] [NORMAL_CLEARING] 2019-05-17 11:22:57.638443 [DEBUG] switch_channel.c:3222 Send signal sofia/external/<calledfaxnumber> [KILL] 2019-05-17 11:22:57.638443 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/<calledfaxnumber> [BREAK] 2019-05-17 11:22:57.638443 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Status changing to 'The call dropped prematurely' This are the most relevant settings on api originate: [DEBUG] switch_event.c:1688 Parsing variable [fax_use_ecm]=[off] [DEBUG] switch_event.c:1688 Parsing variable [absolute_codec_string]=[PCMA@10i] [DEBUG] switch_event.c:1688 Parsing variable [fax_enable_t38]=[false] [DEBUG] switch_event.c:1688 Parsing variable [fax_enable_t38_request]=[false] [DEBUG] switch_event.c:1688 Parsing variable [refuse_t38]=[true] and on the external profile t38-passthru is set to false. -- Ing. Sandro Bordacchini / System Engineering and Product Management sandro.bordacchini(a)nems.it

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FS can't reconnect to PostgreSQL database
by Frédérick Pruneau 29 May '19

29 May '19

Hi guys, I have an issue with FS unable to reconnect to postgresql database. Here is the setup: 2 FreeSWITCH servers with 2 postgresql servers. Each freeswitch is connected to 1 postgresql server. DB servers are synced with BDR. It was working very well until I restarted freeswitch service on one of the servers. Then, I got theses errors showing up on freeswitch console: switch_pgsql.c:384 Query X took too long to complete or database not responding. database is locked against ddl by another node I followed some guides to remove locks with these commands: SELECT * FROM pg_stat_activity; SELECT pg_terminate_backend(${pid}); But the issue is still there. Do you have an idea how to fix this issue? Thanks.

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How to load access control list from redis server of freeswitch?
by shivali chalke 29 May '19

29 May '19

Hi guys, I am trying to find the solution to load the access control list from redis server. I tried to connect to redis server and execute commands to load ACL as well as with hiredis_raw command but didn't get any success. Can anyone help me on this? Thanks in advance.

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