July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How do you determine if DTMFs received are in-band or out-of-band?
by Mark Melling 07 Jul '17

07 Jul '17

Hi, How can you determine if Freeswitch is receiving DTMFs in-band or out-of-band? I'm not entirely clear on terminology here, but is out-of-band the same as rfc 2833? Thanks Mark

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Software Dev Job
by Brian West 07 Jul '17

07 Jul '17

FreeSWITCHers, https://freeswitch-solutions-llc.workable.com/jobs/523308 Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) <https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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PostgreSQL BDR Schema for HA
by Matthew Nichols 07 Jul '17

07 Jul '17

Is there documentation for a PostgreSQL BDR Schema that works with FreeSWITCH HA? The closest I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5 and while the RTP streams transfer over the call/channel state and what not don't, and I get errors like this: 2017-07-07 11:42:04.296420 [DEBUG] switch_pgsql.c:415 Query (insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('eac21399-ecee-40c4-a4e3-9a8f9e4aae7a','outbound','2017-07-07 11:42:04','1499449324','sofia/internal/1002(a)10.1.1.71','CS_INIT','DOWN','XML','default','VFS1','1001','1001','10.1.1.71','1002','XML','default')) returned PGRES_FATAL_ERROR 2017-07-07 11:42:04.296420 [DEBUG] switch_pgsql.c:415 Query (insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('09112fab-0c0f-4ee1-9254-424af3948658','inbound','2017-07-07 11:42:04','1499449324','sofia/internal/1001(a)10.1.1.46','CS_INIT','ACTIVE','XML','default','VFS1','1001','1001','10.1.1.70','1002','XML','default')) returned PGRES_FATAL_ERROR 2017-07-07 11:42:04.296420 [ERR] switch_pgsql.c:656 Error executing query: ERROR: current transaction is aborted, commands ignored until end of transaction block 2017-07-07 11:42:04.496420 [DEBUG] switch_pgsql.c:415 Query (insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('09112fab-0c0f-4ee1-9254-424af3948658','2017-07-07 11:42:04','1499449324','09112fab-0c0f-4ee1-9254-424af3948658','eac21399-ecee-40c4-a4e3-9a8f9e4aae7a','VFS1')) returned PGRES_FATAL_ERROR 2017-07-07 11:42:04.496420 [DEBUG] switch_pgsql.c:415 Query (update channels set read_codec='G722',read_rate='16000',read_bit_rate='64000',write_codec='G722',write_rate='16000',write_bit_rate='64000' where uuid='eac21399-ecee-40c4-a4e3-9a8f9e4aae7a') returned PGRES_FATAL_ERROR 2017-07-07 11:42:04.496420 [ERR] switch_pgsql.c:656 Error executing query: ERROR: current transaction is aborted, commands ignored until end of transaction block It looks like it's trying to insert into the channels/calls tables with the same UUID as before which of course doesn't work since it is a primary key. Does the primary key need to include the hostname as well (and is that only for the calls/channels tables or other tables as well?). Since BDR requires a primary key I'd like to know what FreeSWITCH expects, since it doesn't autogenerate one.

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Verto: no inbound audio when verto client called from conference
by Mark Melling 07 Jul '17

07 Jul '17

Hi, I have a dialplan where a user calls a number which results in them being placed in a conference room, and an outbound called made to a verto client when the conference is created, by defining in the dialplan: <action application="conference_set_auto_outcall" data="user/1000@ $${domain}"/> The verto client (which is basically taken from the verto demo) rings and is answered. The original inbound caller can hear the verto client speaking, but the verto client can't hear the the inbound caller (although the audio from the inbound caller is getting to the conference room). If I change the dialling order - so the verto client makes the inbound call and the SIP client is dialled from the conference using the same technique as before (conference_set_auto_outcall). Then everything appears to work fine, i.e. each caller can here the other. Any suggestions as to what might be the issue or steps I could take to help identify the problem. Thanks Mark

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Controlling tones for RFC2833 DTMFs
by Mark Melling 07 Jul '17

07 Jul '17

Hi, If Freeswitch receives rfc2833 DTMFs, given that these DTMFs are out-of-band, does Freeswitch generate a tone to indicate to the user that a key has been pressed? Assuming it does, is there a way to prevent the tone from being generated, i.e. so the user doesn't hear anything when a key is pressed? Thanks Mark

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SRTP Issue
by John McPherson 07 Jul '17

07 Jul '17

Hello All, *I had the following issue:* https://pastebin.freeswitch.org/view/8722d895#L30 issue is here :[WARNING] switch_core_media.c:4451 Crypto not negotiated but required. This is the log for my issue. The external forwarding does not work when I use rtp_secure_media=true. Everything else works. The only workaround is to put rtp_secure_media=false in the outbound dialplan and make the bridge=loopback instead of outbound. I don't like this workaround. Is there a more elegant solution? I want the bridge=outbound and not loopback *Then I was told that I can use rtp_secure_media=optional* It made me happy as I thought the problem is solved. *But now the problem is rtp_secure_media=optional does not work when the devices only have SRTP_Encryption =ALWAYS or NEVER where the "Optional" is not an option and they cannot receive calls. I know that we can separate inbound and outbound. However, forwarding is both inbound and outbound.* Do you have a solution for me? With my thanks, John McPherson

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freeswitch.Session with LUA
by Stefano Favaro 07 Jul '17

07 Jul '17

Hello, I'm writing a simple script with LUA to connect an inbound call to e new outbound call. Everything work fine but I can't set the ringback tone nor a waiting music on the leg A and also the call_timeout variable. The legA is in silence while the legB is ringing and the call_timeout is always the default 60 seconds. What is wrong with this code? Thanks This is the code: ... legA:streamFile("/usr/share/freeswitch/sounds/wait.wav") legA:setVariable("ringback", "$${uk-ring}") legA:setVariable("call_timeout", 30) legB = freeswitch.Session("{ignore_early_media=true, hangup_after_bridge=true, origination_caller_id_number=1234}sofia/internal/0123456789@myserver) if legB:ready() then freeswitch.bridge(legA, legB) end Stefano Favaro Sviluppo Servizi ed Applicazioni _____________________ Edistar Srl a socio unico soggetta a direzione e coordinamento di YourVoice SpA Via Artigianato 1 – I – 31050 Vedelago (TV) Italy Phone +39 0423 7331 – Fax +39 0423 733133 skype: stefanofavaro www.edistar.com Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potrà essere effettuata denuncia alla competente Autorità. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario è vietata dall’art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation.

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redirect context not working
by Joseph Waite 07 Jul '17

07 Jul '17

Hi Guys I have setup a context called redirect to catch calls sent back from a redirect server with a 300 multiple choices message. However I get the following error when a call goes through. 2017-07-06 23:14:27.403150 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp@185.35.228.51:5060 Standard ROUTING 2017-07-06 23:14:27.403150 [INFO] mod_dialplan_xml.c:637 Processing sipp <sipp>->441554555666 in context redirect 2017-07-06 23:14:27.403150 [INFO] switch_core_state_machine.c:311 No Route, Aborting 2017-07-06 23:14:27.403150 [NOTICE] switch_core_state_machine.c:312 Hangup sofia/external/sipp@185.35.228.51:5060 [CS_ROUTING] [NO_ROUTE_DESTINATION] I have a redirect.xml file as follows in my conf/dialplan <include> <context name="redirect"> <X-PRE-PROCESS cmd="include" data="redirect/*.xml"/> </context> </include> and then a 00_redirect.xml in conf/dialplan/redirect <include> <context name="redirect"> <extension name="Redirect Calls"> <condition> <action application="log" data="INFO Were in the Redirect Context!!"/> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </context> </include> Can anyone give me an idea why the call is not being picked up in the Redirect Calls extension? Regards

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G729 annexb=no not forwarded to leg B
by David Ponzone 07 Jul '17

07 Jul '17

All, If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in the SDP, I would expect annexb=no to be automatically inserted in the SDP to Leg B. It seems either with export absolute_codec_string=G729, and without it (so I send G729 and G711 to Leg B), annexb=no is not sent to Leg B. Is there any parameters or any configuration issue which may lead to annex=no being suppressed (which is an issue, as some carriers default to annexb=yes), or is it the normal behavior, meaning I have to manually export it to leg B if it was requested by A ? Thank you

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running FS issue without make install
by Bipin Patel 06 Jul '17

06 Jul '17

hi, this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu. firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries? secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever -- Regards, Bipin ------------------------------------------------------------------------

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